| Message ID | 20170505193038.32318-1-onemda@gmail.com |
|---|---|
| State | Superseded |
| Headers | show |
On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda@gmail.com> wrote: > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > configure | 2 + > doc/filters.texi | 10 + > libavfilter/Makefile | 1 + > libavfilter/af_afir.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 5 files changed, 498 insertions(+) > create mode 100644 libavfilter/af_afir.c > > diff --git a/configure b/configure > index b3cb5b0..0d83c6a 100755 > --- a/configure > +++ b/configure > @@ -3078,6 +3078,8 @@ unix_protocol_select="network" > # filters > afftfilt_filter_deps="avcodec" > afftfilt_filter_select="fft" > +afir_filter_deps="avcodec" > +afir_filter_select="fft" > amovie_filter_deps="avcodec avformat" > aresample_filter_deps="swresample" > ass_filter_deps="libass" > diff --git a/doc/filters.texi b/doc/filters.texi > index 119e747..ea343d1 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)" > @end example > @end itemize > > +@section afirfilter > + > +Apply an Arbitary Frequency Impulse Response filter. > + > +This filter uses second stream as FIR coefficients. > +If second stream holds single channel, it will be used > +for all input channels in first stream, otherwise > +number of channels in second stream must be same as > +number of channels in first stream. > + > @anchor{aformat} > @section aformat Seems that you forgot to update the documentation. > > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 66c36e4..c797eb5 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o > OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o > OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o > OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o > +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o > OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o > OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o > OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o > diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c > new file mode 100644 > index 0000000..9411c9b > --- /dev/null > +++ b/libavfilter/af_afir.c > @@ -0,0 +1,484 @@ > +/* > + * Copyright (c) 2017 Paul B Mahol > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +/** > + * @file > + * An arbitrary audio FIR filter > + */ > + > +#include "libavutil/audio_fifo.h" > +#include "libavutil/common.h" > +#include "libavutil/opt.h" > +#include "libavcodec/avfft.h" > + > +#include "audio.h" > +#include "avfilter.h" > +#include "formats.h" > +#include "internal.h" > + > +#define MAX_IR_DURATION 20 > + > +typedef struct FIRContext { > + const AVClass *class; > + > + float wet_gain; > + float dry_gain; > + int auto_gain; > + > + float gain; > + > + int eof_coeffs; > + int have_coeffs; > + int nb_coeffs; > + int nb_taps; > + int part_size; > + int nb_partitions; > + int fft_length; > + int nb_channels; > + int nb_coef_channels; > + int one2many; > + int nb_samples; > + > + RDFTContext **rdft, **irdft; > + float **sum; > + float **block; > + FFTComplex **coeff; > + > + AVAudioFifo *fifo[2]; > + AVFrame *in[2]; > + AVFrame *buffer; > + int64_t pts; > +} FIRContext; > + > +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) > +{ > + FIRContext *s = ctx->priv; > + AVFrame *out = arg; > + const FFTComplex *coeff = s->coeff[ch * !s->one2many]; > + const float *src = (const float *)s->in[0]->extended_data[ch]; > + float *dst = (float *)out->extended_data[ch]; > + float *buf = (float *)s->buffer->extended_data[ch]; > + float *sum = s->sum[ch]; > + float *block = s->block[ch]; > + int n, i; > + > + memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1)); > + memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1)); > + for (n = 0; n < s->nb_samples; n++) { > + block[n] = src[n] * s->dry_gain; > + } > + > + av_rdft_calc(s->rdft[ch], block); > + block[s->part_size / 2] = block[1]; > + block[1] = 0; > + > + for (i = 0; i < s->nb_partitions; i++) { > + const int coffset = i * (s->part_size + 1); > + > + for (n = 0; n <= s->part_size; n++) { > + const float re = block[2 * n ]; > + const float im = block[2 * n + 1]; > + const float cre = coeff[coffset + n].re; > + const float cim = coeff[coffset + n].im; > + > + sum[2 * n ] += re * cre - im * cim; > + sum[2 * n + 1] += re * cim + im * cre; > + } > + } > + > + sum[1] = sum[n]; > + av_rdft_calc(s->irdft[ch], sum); > + > + for (n = 0; n < out->nb_samples; n++) { > + float sample; > + > + sample = sum[out->nb_samples + n]; > + dst[n] = sample * s->wet_gain * s->gain; > + buf[n] = sum[n]; > + } > + > + return 0; > +} > + > +static int fir_frame(FIRContext *s, AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + AVFrame *out; > + > + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0])); > + > + out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? s->nb_samples : s->part_size / 2); > + if (!out) > + return AVERROR(ENOMEM); > + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples); > + if (!s->in[0]) { > + av_frame_free(&out); > + return AVERROR(ENOMEM); > + } > + > + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples); > + > + ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); > + > + av_audio_fifo_drain(s->fifo[0], out->nb_samples); > + > + out->pts = s->pts; > + if (s->pts != AV_NOPTS_VALUE) > + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); > + > + av_frame_free(&s->in[0]); > + > + return ff_filter_frame(outlink, out); > +} > + > +static int convert_coeffs(AVFilterContext *ctx) > +{ > + FIRContext *s = ctx->priv; > + int max_nb_taps, i, ch, n, N; > + float power = 0; > + > + s->nb_taps = av_audio_fifo_size(s->fifo[1]); > + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; > + if (s->nb_taps > max_nb_taps) { > + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", s->nb_taps, max_nb_taps); > + return AVERROR(EINVAL); > + } > + > + for (n = 1; (1 << n) < s->nb_taps; n++); > + N = FFMIN(n, 16); > + s->fft_length = 1 << n; > + s->part_size = 1 << (N - 1); > + s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size; > + s->nb_coeffs = s->fft_length + s->nb_partitions; > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->sum[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->sum)); > + if (!s->sum[ch]) > + return AVERROR(ENOMEM); > + } > + > + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { > + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff)); > + if (!s->coeff[ch]) > + return AVERROR(ENOMEM); > + } > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->block[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->block)); > + if (!s->block[ch]) > + return AVERROR(ENOMEM); > + } > + > + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); > + if (!s->buffer) > + return AVERROR(ENOMEM); > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->rdft[ch] = av_rdft_init(N, DFT_R2C); > + s->irdft[ch] = av_rdft_init(N, IDFT_C2R); > + if (!s->rdft[ch] || !s->irdft[ch]) > + return AVERROR(ENOMEM); > + } > + > + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps); > + if (!s->in[1]) > + return AVERROR(ENOMEM); > + > + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps); > + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { > + const float *re = (const float *)s->in[1]->extended_data[!s->one2many * ch]; > + float *block = s->block[ch]; > + FFTComplex *coeff = s->coeff[ch]; > + > + for (i = 0; i < s->nb_partitions; i++) { > + const int offset = i * s->part_size; > + const int coffset = i * (s->part_size + 1); > + const int remaining = s->nb_taps - (i * s->part_size); > + const int size = remaining >= s->part_size ? s->part_size : remaining; > + > + memset(block, 0, sizeof(*block) * (2 * (s->part_size + 1))); > + for (n = 0; n < size; n++) { > + block[n] = re[n + offset]; > + power += block[n] * block[n]; > + } > + > + av_rdft_calc(s->rdft[0], block); > + > + coeff[coffset].re = block[0]; > + coeff[coffset].im = 0; > + for (n = 1; n < s->part_size; n++) { > + coeff[coffset + n].re = block[2 * n]; > + coeff[coffset + n].im = block[2 * n + 1]; > + } > + coeff[coffset + n].re = block[1]; > + coeff[coffset + n].im = 0; > + } > + } > + power /= ctx->inputs[1]->channels; > + > + av_frame_free(&s->in[1]); > + s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : sqrtf(s->part_size)); > + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N); > + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); > + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); > + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions); > + > + s->have_coeffs = 1; > + > + return 0; > +} > + > +static int read_ir(AVFilterLink *link, AVFrame *frame) > +{ > + AVFilterContext *ctx = link->dst; > + FIRContext *s = ctx->priv; > + int nb_taps, max_nb_taps; > + > + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data, > + frame->nb_samples); > + av_frame_free(&frame); > + > + nb_taps = av_audio_fifo_size(s->fifo[1]); > + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; > + if (s->nb_taps > max_nb_taps) { > + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps); > + return AVERROR(EINVAL); > + } > + > + return 0; > +} > + > +static int filter_frame(AVFilterLink *link, AVFrame *frame) > +{ > + AVFilterContext *ctx = link->dst; > + FIRContext *s = ctx->priv; > + AVFilterLink *outlink = ctx->outputs[0]; > + int ret = 0; > + > + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data, > + frame->nb_samples); > + if (s->pts == AV_NOPTS_VALUE) > + s->pts = frame->pts; > + > + av_frame_free(&frame); > + > + if (!s->have_coeffs && s->eof_coeffs) { > + ret = convert_coeffs(ctx); > + if (ret < 0) > + return ret; > + } > + > + if (s->have_coeffs) { > + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) { > + ret = fir_frame(s, outlink); > + if (ret < 0) > + break; > + } > + } > + return ret; > +} > + > +static int request_frame(AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + FIRContext *s = ctx->priv; > + int ret; > + > + if (!s->eof_coeffs) { > + ret = ff_request_frame(ctx->inputs[1]); > + if (ret == AVERROR_EOF) { > + s->eof_coeffs = 1; > + ret = 0; > + } > + return ret; > + } > + ret = ff_request_frame(ctx->inputs[0]); > + if (ret == AVERROR_EOF && s->have_coeffs) { > + while (av_audio_fifo_size(s->fifo[0]) > 0) { > + ret = fir_frame(s, outlink); > + if (ret < 0) > + return ret; > + } > + ret = AVERROR_EOF; > + } > + return ret; > +} > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats; > + AVFilterChannelLayouts *layouts = NULL; > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_FLTP, > + AV_SAMPLE_FMT_NONE > + }; > + int ret, i; > + > + layouts = ff_all_channel_counts(); > + if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0) > + return ret; > + > + for (i = 0; i < 2; i++) { > + layouts = ff_all_channel_counts(); > + if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0) > + return ret; > + } > + > + formats = ff_make_format_list(sample_fmts); > + if ((ret = ff_set_common_formats(ctx, formats)) < 0) > + return ret; > + > + formats = ff_all_samplerates(); > + return ff_set_common_samplerates(ctx, formats); > +} > + > +static int config_output(AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + FIRContext *s = ctx->priv; > + > + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels && > + ctx->inputs[1]->channels != 1) { > + av_log(ctx, AV_LOG_ERROR, > + "Second input must have same number of channels as first input or " > + "exactly 1 channel.\n"); > + return AVERROR(EINVAL); > + } > + > + s->one2many = ctx->inputs[1]->channels == 1; > + outlink->sample_rate = ctx->inputs[0]->sample_rate; > + outlink->time_base = ctx->inputs[0]->time_base; > + outlink->channel_layout = ctx->inputs[0]->channel_layout; > + outlink->channels = ctx->inputs[0]->channels; > + > + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024); > + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024); > + if (!s->fifo[0] || !s->fifo[1]) > + return AVERROR(ENOMEM); > + > + s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); > + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); > + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); > + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); > + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); > + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) > + return AVERROR(ENOMEM); > + > + s->nb_channels = outlink->channels; > + s->nb_coef_channels = ctx->inputs[1]->channels; > + s->pts = AV_NOPTS_VALUE; > + > + return 0; > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + FIRContext *s = ctx->priv; > + int ch; > + > + if (s->sum) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_freep(&s->sum[ch]); > + } > + } > + av_freep(&s->sum); > + > + if (s->coeff) { > + for (ch = 0; ch < s->nb_coef_channels; ch++) { > + av_freep(&s->coeff[ch]); > + } > + } > + av_freep(&s->coeff); > + > + if (s->block) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_freep(&s->block[ch]); > + } > + } > + av_freep(&s->block); > + > + if (s->rdft) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_rdft_end(s->rdft[ch]); > + } > + } > + av_freep(&s->rdft); > + > + if (s->irdft) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_rdft_end(s->irdft[ch]); > + } > + } > + av_freep(&s->irdft); > + > + av_frame_free(&s->in[0]); > + av_frame_free(&s->in[1]); > + av_frame_free(&s->buffer); > + > + av_audio_fifo_free(s->fifo[0]); > + av_audio_fifo_free(s->fifo[1]); > +} > + > +static const AVFilterPad afir_inputs[] = { > + { > + .name = "main", > + .type = AVMEDIA_TYPE_AUDIO, > + .filter_frame = filter_frame, > + },{ > + .name = "ir", > + .type = AVMEDIA_TYPE_AUDIO, > + .filter_frame = read_ir, > + }, > + { NULL } > +}; > + > +static const AVFilterPad afir_outputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + .config_props = config_output, > + .request_frame = request_frame, > + }, > + { NULL } > +}; > + > +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM > +#define OFFSET(x) offsetof(FIRContext, x) > + > +static const AVOption afir_options[] = { > + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, > + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, > + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, > + { NULL } > +}; > + > +AVFILTER_DEFINE_CLASS(afir); > + > +AVFilter ff_af_afir = { > + .name = "afir", > + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."), > + .priv_size = sizeof(FIRContext), > + .priv_class = &afir_class, > + .query_formats = query_formats, > + .uninit = uninit, > + .inputs = afir_inputs, > + .outputs = afir_outputs, > + .flags = AVFILTER_FLAG_SLICE_THREADS, > +}; > diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c > index 8fb87eb..555c442 100644 > --- a/libavfilter/allfilters.c > +++ b/libavfilter/allfilters.c > @@ -50,6 +50,7 @@ static void register_all(void) > REGISTER_FILTER(AEVAL, aeval, af); > REGISTER_FILTER(AFADE, afade, af); > REGISTER_FILTER(AFFTFILT, afftfilt, af); > + REGISTER_FILTER(AFIR, afir, af); > REGISTER_FILTER(AFORMAT, aformat, af); > REGISTER_FILTER(AGATE, agate, af); > REGISTER_FILTER(AINTERLEAVE, ainterleave, af); Seems that the partitioned convolution code here doesn't work. I can't help here. IMHO, you should stuck to traditional convolution code. Thank's
On 5/6/17, Muhammad Faiz <mfcc64@gmail.com> wrote: > On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda@gmail.com> wrote: >> Signed-off-by: Paul B Mahol <onemda@gmail.com> >> --- >> configure | 2 + >> doc/filters.texi | 10 + >> libavfilter/Makefile | 1 + >> libavfilter/af_afir.c | 484 >> +++++++++++++++++++++++++++++++++++++++++++++++ >> libavfilter/allfilters.c | 1 + >> 5 files changed, 498 insertions(+) >> create mode 100644 libavfilter/af_afir.c >> >> diff --git a/configure b/configure >> index b3cb5b0..0d83c6a 100755 >> --- a/configure >> +++ b/configure >> @@ -3078,6 +3078,8 @@ unix_protocol_select="network" >> # filters >> afftfilt_filter_deps="avcodec" >> afftfilt_filter_select="fft" >> +afir_filter_deps="avcodec" >> +afir_filter_select="fft" >> amovie_filter_deps="avcodec avformat" >> aresample_filter_deps="swresample" >> ass_filter_deps="libass" >> diff --git a/doc/filters.texi b/doc/filters.texi >> index 119e747..ea343d1 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)" >> @end example >> @end itemize >> >> +@section afirfilter >> + >> +Apply an Arbitary Frequency Impulse Response filter. >> + >> +This filter uses second stream as FIR coefficients. >> +If second stream holds single channel, it will be used >> +for all input channels in first stream, otherwise >> +number of channels in second stream must be same as >> +number of channels in first stream. >> + >> @anchor{aformat} >> @section aformat > > Seems that you forgot to update the documentation. > >> >> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >> index 66c36e4..c797eb5 100644 >> --- a/libavfilter/Makefile >> +++ b/libavfilter/Makefile >> @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += >> af_aemphasis.o >> OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o >> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o >> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o >> window_func.o >> +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o >> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o >> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o >> OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o >> diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c >> new file mode 100644 >> index 0000000..9411c9b >> --- /dev/null >> +++ b/libavfilter/af_afir.c >> @@ -0,0 +1,484 @@ >> +/* >> + * Copyright (c) 2017 Paul B Mahol >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA >> 02110-1301 USA >> + */ >> + >> +/** >> + * @file >> + * An arbitrary audio FIR filter >> + */ >> + >> +#include "libavutil/audio_fifo.h" >> +#include "libavutil/common.h" >> +#include "libavutil/opt.h" >> +#include "libavcodec/avfft.h" >> + >> +#include "audio.h" >> +#include "avfilter.h" >> +#include "formats.h" >> +#include "internal.h" >> + >> +#define MAX_IR_DURATION 20 >> + >> +typedef struct FIRContext { >> + const AVClass *class; >> + >> + float wet_gain; >> + float dry_gain; >> + int auto_gain; >> + >> + float gain; >> + >> + int eof_coeffs; >> + int have_coeffs; >> + int nb_coeffs; >> + int nb_taps; >> + int part_size; >> + int nb_partitions; >> + int fft_length; >> + int nb_channels; >> + int nb_coef_channels; >> + int one2many; >> + int nb_samples; >> + >> + RDFTContext **rdft, **irdft; >> + float **sum; >> + float **block; >> + FFTComplex **coeff; >> + >> + AVAudioFifo *fifo[2]; >> + AVFrame *in[2]; >> + AVFrame *buffer; >> + int64_t pts; >> +} FIRContext; >> + >> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int >> nb_jobs) >> +{ >> + FIRContext *s = ctx->priv; >> + AVFrame *out = arg; >> + const FFTComplex *coeff = s->coeff[ch * !s->one2many]; >> + const float *src = (const float *)s->in[0]->extended_data[ch]; >> + float *dst = (float *)out->extended_data[ch]; >> + float *buf = (float *)s->buffer->extended_data[ch]; >> + float *sum = s->sum[ch]; >> + float *block = s->block[ch]; >> + int n, i; >> + >> + memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1)); >> + memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1)); >> + for (n = 0; n < s->nb_samples; n++) { >> + block[n] = src[n] * s->dry_gain; >> + } >> + >> + av_rdft_calc(s->rdft[ch], block); >> + block[s->part_size / 2] = block[1]; >> + block[1] = 0; >> + >> + for (i = 0; i < s->nb_partitions; i++) { >> + const int coffset = i * (s->part_size + 1); >> + >> + for (n = 0; n <= s->part_size; n++) { >> + const float re = block[2 * n ]; >> + const float im = block[2 * n + 1]; >> + const float cre = coeff[coffset + n].re; >> + const float cim = coeff[coffset + n].im; >> + >> + sum[2 * n ] += re * cre - im * cim; >> + sum[2 * n + 1] += re * cim + im * cre; >> + } >> + } >> + >> + sum[1] = sum[n]; >> + av_rdft_calc(s->irdft[ch], sum); >> + >> + for (n = 0; n < out->nb_samples; n++) { >> + float sample; >> + >> + sample = sum[out->nb_samples + n]; >> + dst[n] = sample * s->wet_gain * s->gain; >> + buf[n] = sum[n]; >> + } >> + >> + return 0; >> +} >> + >> +static int fir_frame(FIRContext *s, AVFilterLink *outlink) >> +{ >> + AVFilterContext *ctx = outlink->src; >> + AVFrame *out; >> + >> + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0])); >> + >> + out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? >> s->nb_samples : s->part_size / 2); >> + if (!out) >> + return AVERROR(ENOMEM); >> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples); >> + if (!s->in[0]) { >> + av_frame_free(&out); >> + return AVERROR(ENOMEM); >> + } >> + >> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, >> s->nb_samples); >> + >> + ctx->internal->execute(ctx, fir_channel, out, NULL, >> outlink->channels); >> + >> + av_audio_fifo_drain(s->fifo[0], out->nb_samples); >> + >> + out->pts = s->pts; >> + if (s->pts != AV_NOPTS_VALUE) >> + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, >> outlink->sample_rate}, outlink->time_base); >> + >> + av_frame_free(&s->in[0]); >> + >> + return ff_filter_frame(outlink, out); >> +} >> + >> +static int convert_coeffs(AVFilterContext *ctx) >> +{ >> + FIRContext *s = ctx->priv; >> + int max_nb_taps, i, ch, n, N; >> + float power = 0; >> + >> + s->nb_taps = av_audio_fifo_size(s->fifo[1]); >> + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; >> + if (s->nb_taps > max_nb_taps) { >> + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > >> %d.\n", s->nb_taps, max_nb_taps); >> + return AVERROR(EINVAL); >> + } >> + >> + for (n = 1; (1 << n) < s->nb_taps; n++); >> + N = FFMIN(n, 16); >> + s->fft_length = 1 << n; >> + s->part_size = 1 << (N - 1); >> + s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size; >> + s->nb_coeffs = s->fft_length + s->nb_partitions; >> + >> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >> + s->sum[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->sum)); >> + if (!s->sum[ch]) >> + return AVERROR(ENOMEM); >> + } >> + >> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { >> + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff)); >> + if (!s->coeff[ch]) >> + return AVERROR(ENOMEM); >> + } >> + >> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >> + s->block[ch] = av_calloc(2 * (s->part_size + 1), >> sizeof(**s->block)); >> + if (!s->block[ch]) >> + return AVERROR(ENOMEM); >> + } >> + >> + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); >> + if (!s->buffer) >> + return AVERROR(ENOMEM); >> + >> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >> + s->rdft[ch] = av_rdft_init(N, DFT_R2C); >> + s->irdft[ch] = av_rdft_init(N, IDFT_C2R); >> + if (!s->rdft[ch] || !s->irdft[ch]) >> + return AVERROR(ENOMEM); >> + } >> + >> + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps); >> + if (!s->in[1]) >> + return AVERROR(ENOMEM); >> + >> + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, >> s->nb_taps); >> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { >> + const float *re = (const float >> *)s->in[1]->extended_data[!s->one2many * ch]; >> + float *block = s->block[ch]; >> + FFTComplex *coeff = s->coeff[ch]; >> + >> + for (i = 0; i < s->nb_partitions; i++) { >> + const int offset = i * s->part_size; >> + const int coffset = i * (s->part_size + 1); >> + const int remaining = s->nb_taps - (i * s->part_size); >> + const int size = remaining >= s->part_size ? s->part_size : >> remaining; >> + >> + memset(block, 0, sizeof(*block) * (2 * (s->part_size + 1))); >> + for (n = 0; n < size; n++) { >> + block[n] = re[n + offset]; >> + power += block[n] * block[n]; >> + } >> + >> + av_rdft_calc(s->rdft[0], block); >> + >> + coeff[coffset].re = block[0]; >> + coeff[coffset].im = 0; >> + for (n = 1; n < s->part_size; n++) { >> + coeff[coffset + n].re = block[2 * n]; >> + coeff[coffset + n].im = block[2 * n + 1]; >> + } >> + coeff[coffset + n].re = block[1]; >> + coeff[coffset + n].im = 0; >> + } >> + } >> + power /= ctx->inputs[1]->channels; >> + >> + av_frame_free(&s->in[1]); >> + s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : >> sqrtf(s->part_size)); >> + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N); >> + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); >> + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); >> + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions); >> + >> + s->have_coeffs = 1; >> + >> + return 0; >> +} >> + >> +static int read_ir(AVFilterLink *link, AVFrame *frame) >> +{ >> + AVFilterContext *ctx = link->dst; >> + FIRContext *s = ctx->priv; >> + int nb_taps, max_nb_taps; >> + >> + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data, >> + frame->nb_samples); >> + av_frame_free(&frame); >> + >> + nb_taps = av_audio_fifo_size(s->fifo[1]); >> + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; >> + if (s->nb_taps > max_nb_taps) { >> + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > >> %d.\n", nb_taps, max_nb_taps); >> + return AVERROR(EINVAL); >> + } >> + >> + return 0; >> +} >> + >> +static int filter_frame(AVFilterLink *link, AVFrame *frame) >> +{ >> + AVFilterContext *ctx = link->dst; >> + FIRContext *s = ctx->priv; >> + AVFilterLink *outlink = ctx->outputs[0]; >> + int ret = 0; >> + >> + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data, >> + frame->nb_samples); >> + if (s->pts == AV_NOPTS_VALUE) >> + s->pts = frame->pts; >> + >> + av_frame_free(&frame); >> + >> + if (!s->have_coeffs && s->eof_coeffs) { >> + ret = convert_coeffs(ctx); >> + if (ret < 0) >> + return ret; >> + } >> + >> + if (s->have_coeffs) { >> + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) { >> + ret = fir_frame(s, outlink); >> + if (ret < 0) >> + break; >> + } >> + } >> + return ret; >> +} >> + >> +static int request_frame(AVFilterLink *outlink) >> +{ >> + AVFilterContext *ctx = outlink->src; >> + FIRContext *s = ctx->priv; >> + int ret; >> + >> + if (!s->eof_coeffs) { >> + ret = ff_request_frame(ctx->inputs[1]); >> + if (ret == AVERROR_EOF) { >> + s->eof_coeffs = 1; >> + ret = 0; >> + } >> + return ret; >> + } >> + ret = ff_request_frame(ctx->inputs[0]); >> + if (ret == AVERROR_EOF && s->have_coeffs) { >> + while (av_audio_fifo_size(s->fifo[0]) > 0) { >> + ret = fir_frame(s, outlink); >> + if (ret < 0) >> + return ret; >> + } >> + ret = AVERROR_EOF; >> + } >> + return ret; >> +} >> + >> +static int query_formats(AVFilterContext *ctx) >> +{ >> + AVFilterFormats *formats; >> + AVFilterChannelLayouts *layouts = NULL; >> + static const enum AVSampleFormat sample_fmts[] = { >> + AV_SAMPLE_FMT_FLTP, >> + AV_SAMPLE_FMT_NONE >> + }; >> + int ret, i; >> + >> + layouts = ff_all_channel_counts(); >> + if ((ret = ff_channel_layouts_ref(layouts, >> &ctx->outputs[0]->in_channel_layouts)) < 0) >> + return ret; >> + >> + for (i = 0; i < 2; i++) { >> + layouts = ff_all_channel_counts(); >> + if ((ret = ff_channel_layouts_ref(layouts, >> &ctx->inputs[i]->out_channel_layouts)) < 0) >> + return ret; >> + } >> + >> + formats = ff_make_format_list(sample_fmts); >> + if ((ret = ff_set_common_formats(ctx, formats)) < 0) >> + return ret; >> + >> + formats = ff_all_samplerates(); >> + return ff_set_common_samplerates(ctx, formats); >> +} >> + >> +static int config_output(AVFilterLink *outlink) >> +{ >> + AVFilterContext *ctx = outlink->src; >> + FIRContext *s = ctx->priv; >> + >> + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels && >> + ctx->inputs[1]->channels != 1) { >> + av_log(ctx, AV_LOG_ERROR, >> + "Second input must have same number of channels as first >> input or " >> + "exactly 1 channel.\n"); >> + return AVERROR(EINVAL); >> + } >> + >> + s->one2many = ctx->inputs[1]->channels == 1; >> + outlink->sample_rate = ctx->inputs[0]->sample_rate; >> + outlink->time_base = ctx->inputs[0]->time_base; >> + outlink->channel_layout = ctx->inputs[0]->channel_layout; >> + outlink->channels = ctx->inputs[0]->channels; >> + >> + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, >> ctx->inputs[0]->channels, 1024); >> + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, >> ctx->inputs[1]->channels, 1024); >> + if (!s->fifo[0] || !s->fifo[1]) >> + return AVERROR(ENOMEM); >> + >> + s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); >> + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); >> + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); >> + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); >> + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); >> + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) >> + return AVERROR(ENOMEM); >> + >> + s->nb_channels = outlink->channels; >> + s->nb_coef_channels = ctx->inputs[1]->channels; >> + s->pts = AV_NOPTS_VALUE; >> + >> + return 0; >> +} >> + >> +static av_cold void uninit(AVFilterContext *ctx) >> +{ >> + FIRContext *s = ctx->priv; >> + int ch; >> + >> + if (s->sum) { >> + for (ch = 0; ch < s->nb_channels; ch++) { >> + av_freep(&s->sum[ch]); >> + } >> + } >> + av_freep(&s->sum); >> + >> + if (s->coeff) { >> + for (ch = 0; ch < s->nb_coef_channels; ch++) { >> + av_freep(&s->coeff[ch]); >> + } >> + } >> + av_freep(&s->coeff); >> + >> + if (s->block) { >> + for (ch = 0; ch < s->nb_channels; ch++) { >> + av_freep(&s->block[ch]); >> + } >> + } >> + av_freep(&s->block); >> + >> + if (s->rdft) { >> + for (ch = 0; ch < s->nb_channels; ch++) { >> + av_rdft_end(s->rdft[ch]); >> + } >> + } >> + av_freep(&s->rdft); >> + >> + if (s->irdft) { >> + for (ch = 0; ch < s->nb_channels; ch++) { >> + av_rdft_end(s->irdft[ch]); >> + } >> + } >> + av_freep(&s->irdft); >> + >> + av_frame_free(&s->in[0]); >> + av_frame_free(&s->in[1]); >> + av_frame_free(&s->buffer); >> + >> + av_audio_fifo_free(s->fifo[0]); >> + av_audio_fifo_free(s->fifo[1]); >> +} >> + >> +static const AVFilterPad afir_inputs[] = { >> + { >> + .name = "main", >> + .type = AVMEDIA_TYPE_AUDIO, >> + .filter_frame = filter_frame, >> + },{ >> + .name = "ir", >> + .type = AVMEDIA_TYPE_AUDIO, >> + .filter_frame = read_ir, >> + }, >> + { NULL } >> +}; >> + >> +static const AVFilterPad afir_outputs[] = { >> + { >> + .name = "default", >> + .type = AVMEDIA_TYPE_AUDIO, >> + .config_props = config_output, >> + .request_frame = request_frame, >> + }, >> + { NULL } >> +}; >> + >> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >> +#define OFFSET(x) offsetof(FIRContext, x) >> + >> +static const AVOption afir_options[] = { >> + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, >> {.dbl=1}, 0, 1, AF }, >> + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, >> {.dbl=1}, 0, 1, AF }, >> + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, >> {.i64=1}, 0, 1, AF }, >> + { NULL } >> +}; >> + >> +AVFILTER_DEFINE_CLASS(afir); >> + >> +AVFilter ff_af_afir = { >> + .name = "afir", >> + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response >> filter with supplied coefficients in 2nd stream."), >> + .priv_size = sizeof(FIRContext), >> + .priv_class = &afir_class, >> + .query_formats = query_formats, >> + .uninit = uninit, >> + .inputs = afir_inputs, >> + .outputs = afir_outputs, >> + .flags = AVFILTER_FLAG_SLICE_THREADS, >> +}; >> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >> index 8fb87eb..555c442 100644 >> --- a/libavfilter/allfilters.c >> +++ b/libavfilter/allfilters.c >> @@ -50,6 +50,7 @@ static void register_all(void) >> REGISTER_FILTER(AEVAL, aeval, af); >> REGISTER_FILTER(AFADE, afade, af); >> REGISTER_FILTER(AFFTFILT, afftfilt, af); >> + REGISTER_FILTER(AFIR, afir, af); >> REGISTER_FILTER(AFORMAT, aformat, af); >> REGISTER_FILTER(AGATE, agate, af); >> REGISTER_FILTER(AINTERLEAVE, ainterleave, af); > > Seems that the partitioned convolution code here doesn't work. I can't > help here. > IMHO, you should stuck to traditional convolution code. Never, because non-partitioned OLA/OLS is very limited in usage, and thus considered useless.
On Sat, May 6, 2017 at 3:54 PM, Paul B Mahol <onemda@gmail.com> wrote: > On 5/6/17, Muhammad Faiz <mfcc64@gmail.com> wrote: >> On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda@gmail.com> wrote: >>> Signed-off-by: Paul B Mahol <onemda@gmail.com> >>> --- >>> configure | 2 + >>> doc/filters.texi | 10 + >>> libavfilter/Makefile | 1 + >>> libavfilter/af_afir.c | 484 >>> +++++++++++++++++++++++++++++++++++++++++++++++ >>> libavfilter/allfilters.c | 1 + >>> 5 files changed, 498 insertions(+) >>> create mode 100644 libavfilter/af_afir.c >>> >>> diff --git a/configure b/configure >>> index b3cb5b0..0d83c6a 100755 >>> --- a/configure >>> +++ b/configure >>> @@ -3078,6 +3078,8 @@ unix_protocol_select="network" >>> # filters >>> afftfilt_filter_deps="avcodec" >>> afftfilt_filter_select="fft" >>> +afir_filter_deps="avcodec" >>> +afir_filter_select="fft" >>> amovie_filter_deps="avcodec avformat" >>> aresample_filter_deps="swresample" >>> ass_filter_deps="libass" >>> diff --git a/doc/filters.texi b/doc/filters.texi >>> index 119e747..ea343d1 100644 >>> --- a/doc/filters.texi >>> +++ b/doc/filters.texi >>> @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)" >>> @end example >>> @end itemize >>> >>> +@section afirfilter >>> + >>> +Apply an Arbitary Frequency Impulse Response filter. >>> + >>> +This filter uses second stream as FIR coefficients. >>> +If second stream holds single channel, it will be used >>> +for all input channels in first stream, otherwise >>> +number of channels in second stream must be same as >>> +number of channels in first stream. >>> + >>> @anchor{aformat} >>> @section aformat >> >> Seems that you forgot to update the documentation. >> >>> >>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >>> index 66c36e4..c797eb5 100644 >>> --- a/libavfilter/Makefile >>> +++ b/libavfilter/Makefile >>> @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += >>> af_aemphasis.o >>> OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o >>> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o >>> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o >>> window_func.o >>> +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o >>> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o >>> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o >>> OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o >>> diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c >>> new file mode 100644 >>> index 0000000..9411c9b >>> --- /dev/null >>> +++ b/libavfilter/af_afir.c >>> @@ -0,0 +1,484 @@ >>> +/* >>> + * Copyright (c) 2017 Paul B Mahol >>> + * >>> + * This file is part of FFmpeg. >>> + * >>> + * FFmpeg is free software; you can redistribute it and/or >>> + * modify it under the terms of the GNU Lesser General Public >>> + * License as published by the Free Software Foundation; either >>> + * version 2.1 of the License, or (at your option) any later version. >>> + * >>> + * FFmpeg is distributed in the hope that it will be useful, >>> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >>> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >>> + * Lesser General Public License for more details. >>> + * >>> + * You should have received a copy of the GNU Lesser General Public >>> + * License along with FFmpeg; if not, write to the Free Software >>> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA >>> 02110-1301 USA >>> + */ >>> + >>> +/** >>> + * @file >>> + * An arbitrary audio FIR filter >>> + */ >>> + >>> +#include "libavutil/audio_fifo.h" >>> +#include "libavutil/common.h" >>> +#include "libavutil/opt.h" >>> +#include "libavcodec/avfft.h" >>> + >>> +#include "audio.h" >>> +#include "avfilter.h" >>> +#include "formats.h" >>> +#include "internal.h" >>> + >>> +#define MAX_IR_DURATION 20 >>> + >>> +typedef struct FIRContext { >>> + const AVClass *class; >>> + >>> + float wet_gain; >>> + float dry_gain; >>> + int auto_gain; >>> + >>> + float gain; >>> + >>> + int eof_coeffs; >>> + int have_coeffs; >>> + int nb_coeffs; >>> + int nb_taps; >>> + int part_size; >>> + int nb_partitions; >>> + int fft_length; >>> + int nb_channels; >>> + int nb_coef_channels; >>> + int one2many; >>> + int nb_samples; >>> + >>> + RDFTContext **rdft, **irdft; >>> + float **sum; >>> + float **block; >>> + FFTComplex **coeff; >>> + >>> + AVAudioFifo *fifo[2]; >>> + AVFrame *in[2]; >>> + AVFrame *buffer; >>> + int64_t pts; >>> +} FIRContext; >>> + >>> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int >>> nb_jobs) >>> +{ >>> + FIRContext *s = ctx->priv; >>> + AVFrame *out = arg; >>> + const FFTComplex *coeff = s->coeff[ch * !s->one2many]; >>> + const float *src = (const float *)s->in[0]->extended_data[ch]; >>> + float *dst = (float *)out->extended_data[ch]; >>> + float *buf = (float *)s->buffer->extended_data[ch]; >>> + float *sum = s->sum[ch]; >>> + float *block = s->block[ch]; >>> + int n, i; >>> + >>> + memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1)); >>> + memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1)); >>> + for (n = 0; n < s->nb_samples; n++) { >>> + block[n] = src[n] * s->dry_gain; >>> + } >>> + >>> + av_rdft_calc(s->rdft[ch], block); >>> + block[s->part_size / 2] = block[1]; >>> + block[1] = 0; >>> + >>> + for (i = 0; i < s->nb_partitions; i++) { >>> + const int coffset = i * (s->part_size + 1); >>> + >>> + for (n = 0; n <= s->part_size; n++) { >>> + const float re = block[2 * n ]; >>> + const float im = block[2 * n + 1]; >>> + const float cre = coeff[coffset + n].re; >>> + const float cim = coeff[coffset + n].im; >>> + >>> + sum[2 * n ] += re * cre - im * cim; >>> + sum[2 * n + 1] += re * cim + im * cre; >>> + } >>> + } >>> + >>> + sum[1] = sum[n]; >>> + av_rdft_calc(s->irdft[ch], sum); >>> + >>> + for (n = 0; n < out->nb_samples; n++) { >>> + float sample; >>> + >>> + sample = sum[out->nb_samples + n]; >>> + dst[n] = sample * s->wet_gain * s->gain; >>> + buf[n] = sum[n]; >>> + } >>> + >>> + return 0; >>> +} >>> + >>> +static int fir_frame(FIRContext *s, AVFilterLink *outlink) >>> +{ >>> + AVFilterContext *ctx = outlink->src; >>> + AVFrame *out; >>> + >>> + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0])); >>> + >>> + out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? >>> s->nb_samples : s->part_size / 2); >>> + if (!out) >>> + return AVERROR(ENOMEM); >>> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples); >>> + if (!s->in[0]) { >>> + av_frame_free(&out); >>> + return AVERROR(ENOMEM); >>> + } >>> + >>> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, >>> s->nb_samples); >>> + >>> + ctx->internal->execute(ctx, fir_channel, out, NULL, >>> outlink->channels); >>> + >>> + av_audio_fifo_drain(s->fifo[0], out->nb_samples); >>> + >>> + out->pts = s->pts; >>> + if (s->pts != AV_NOPTS_VALUE) >>> + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, >>> outlink->sample_rate}, outlink->time_base); >>> + >>> + av_frame_free(&s->in[0]); >>> + >>> + return ff_filter_frame(outlink, out); >>> +} >>> + >>> +static int convert_coeffs(AVFilterContext *ctx) >>> +{ >>> + FIRContext *s = ctx->priv; >>> + int max_nb_taps, i, ch, n, N; >>> + float power = 0; >>> + >>> + s->nb_taps = av_audio_fifo_size(s->fifo[1]); >>> + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; >>> + if (s->nb_taps > max_nb_taps) { >>> + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > >>> %d.\n", s->nb_taps, max_nb_taps); >>> + return AVERROR(EINVAL); >>> + } >>> + >>> + for (n = 1; (1 << n) < s->nb_taps; n++); >>> + N = FFMIN(n, 16); >>> + s->fft_length = 1 << n; >>> + s->part_size = 1 << (N - 1); >>> + s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size; >>> + s->nb_coeffs = s->fft_length + s->nb_partitions; >>> + >>> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >>> + s->sum[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->sum)); >>> + if (!s->sum[ch]) >>> + return AVERROR(ENOMEM); >>> + } >>> + >>> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { >>> + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff)); >>> + if (!s->coeff[ch]) >>> + return AVERROR(ENOMEM); >>> + } >>> + >>> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >>> + s->block[ch] = av_calloc(2 * (s->part_size + 1), >>> sizeof(**s->block)); >>> + if (!s->block[ch]) >>> + return AVERROR(ENOMEM); >>> + } >>> + >>> + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); >>> + if (!s->buffer) >>> + return AVERROR(ENOMEM); >>> + >>> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { >>> + s->rdft[ch] = av_rdft_init(N, DFT_R2C); >>> + s->irdft[ch] = av_rdft_init(N, IDFT_C2R); >>> + if (!s->rdft[ch] || !s->irdft[ch]) >>> + return AVERROR(ENOMEM); >>> + } >>> + >>> + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps); >>> + if (!s->in[1]) >>> + return AVERROR(ENOMEM); >>> + >>> + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, >>> s->nb_taps); >>> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { >>> + const float *re = (const float >>> *)s->in[1]->extended_data[!s->one2many * ch]; >>> + float *block = s->block[ch]; >>> + FFTComplex *coeff = s->coeff[ch]; >>> + >>> + for (i = 0; i < s->nb_partitions; i++) { >>> + const int offset = i * s->part_size; >>> + const int coffset = i * (s->part_size + 1); >>> + const int remaining = s->nb_taps - (i * s->part_size); >>> + const int size = remaining >= s->part_size ? s->part_size : >>> remaining; >>> + >>> + memset(block, 0, sizeof(*block) * (2 * (s->part_size + 1))); >>> + for (n = 0; n < size; n++) { >>> + block[n] = re[n + offset]; >>> + power += block[n] * block[n]; >>> + } >>> + >>> + av_rdft_calc(s->rdft[0], block); >>> + >>> + coeff[coffset].re = block[0]; >>> + coeff[coffset].im = 0; >>> + for (n = 1; n < s->part_size; n++) { >>> + coeff[coffset + n].re = block[2 * n]; >>> + coeff[coffset + n].im = block[2 * n + 1]; >>> + } >>> + coeff[coffset + n].re = block[1]; >>> + coeff[coffset + n].im = 0; >>> + } >>> + } >>> + power /= ctx->inputs[1]->channels; >>> + >>> + av_frame_free(&s->in[1]); >>> + s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : >>> sqrtf(s->part_size)); >>> + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N); >>> + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); >>> + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); >>> + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions); >>> + >>> + s->have_coeffs = 1; >>> + >>> + return 0; >>> +} >>> + >>> +static int read_ir(AVFilterLink *link, AVFrame *frame) >>> +{ >>> + AVFilterContext *ctx = link->dst; >>> + FIRContext *s = ctx->priv; >>> + int nb_taps, max_nb_taps; >>> + >>> + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data, >>> + frame->nb_samples); >>> + av_frame_free(&frame); >>> + >>> + nb_taps = av_audio_fifo_size(s->fifo[1]); >>> + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; >>> + if (s->nb_taps > max_nb_taps) { >>> + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > >>> %d.\n", nb_taps, max_nb_taps); >>> + return AVERROR(EINVAL); >>> + } >>> + >>> + return 0; >>> +} >>> + >>> +static int filter_frame(AVFilterLink *link, AVFrame *frame) >>> +{ >>> + AVFilterContext *ctx = link->dst; >>> + FIRContext *s = ctx->priv; >>> + AVFilterLink *outlink = ctx->outputs[0]; >>> + int ret = 0; >>> + >>> + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data, >>> + frame->nb_samples); >>> + if (s->pts == AV_NOPTS_VALUE) >>> + s->pts = frame->pts; >>> + >>> + av_frame_free(&frame); >>> + >>> + if (!s->have_coeffs && s->eof_coeffs) { >>> + ret = convert_coeffs(ctx); >>> + if (ret < 0) >>> + return ret; >>> + } >>> + >>> + if (s->have_coeffs) { >>> + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) { >>> + ret = fir_frame(s, outlink); >>> + if (ret < 0) >>> + break; >>> + } >>> + } >>> + return ret; >>> +} >>> + >>> +static int request_frame(AVFilterLink *outlink) >>> +{ >>> + AVFilterContext *ctx = outlink->src; >>> + FIRContext *s = ctx->priv; >>> + int ret; >>> + >>> + if (!s->eof_coeffs) { >>> + ret = ff_request_frame(ctx->inputs[1]); >>> + if (ret == AVERROR_EOF) { >>> + s->eof_coeffs = 1; >>> + ret = 0; >>> + } >>> + return ret; >>> + } >>> + ret = ff_request_frame(ctx->inputs[0]); >>> + if (ret == AVERROR_EOF && s->have_coeffs) { >>> + while (av_audio_fifo_size(s->fifo[0]) > 0) { >>> + ret = fir_frame(s, outlink); >>> + if (ret < 0) >>> + return ret; >>> + } >>> + ret = AVERROR_EOF; >>> + } >>> + return ret; >>> +} >>> + >>> +static int query_formats(AVFilterContext *ctx) >>> +{ >>> + AVFilterFormats *formats; >>> + AVFilterChannelLayouts *layouts = NULL; >>> + static const enum AVSampleFormat sample_fmts[] = { >>> + AV_SAMPLE_FMT_FLTP, >>> + AV_SAMPLE_FMT_NONE >>> + }; >>> + int ret, i; >>> + >>> + layouts = ff_all_channel_counts(); >>> + if ((ret = ff_channel_layouts_ref(layouts, >>> &ctx->outputs[0]->in_channel_layouts)) < 0) >>> + return ret; >>> + >>> + for (i = 0; i < 2; i++) { >>> + layouts = ff_all_channel_counts(); >>> + if ((ret = ff_channel_layouts_ref(layouts, >>> &ctx->inputs[i]->out_channel_layouts)) < 0) >>> + return ret; >>> + } >>> + >>> + formats = ff_make_format_list(sample_fmts); >>> + if ((ret = ff_set_common_formats(ctx, formats)) < 0) >>> + return ret; >>> + >>> + formats = ff_all_samplerates(); >>> + return ff_set_common_samplerates(ctx, formats); >>> +} >>> + >>> +static int config_output(AVFilterLink *outlink) >>> +{ >>> + AVFilterContext *ctx = outlink->src; >>> + FIRContext *s = ctx->priv; >>> + >>> + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels && >>> + ctx->inputs[1]->channels != 1) { >>> + av_log(ctx, AV_LOG_ERROR, >>> + "Second input must have same number of channels as first >>> input or " >>> + "exactly 1 channel.\n"); >>> + return AVERROR(EINVAL); >>> + } >>> + >>> + s->one2many = ctx->inputs[1]->channels == 1; >>> + outlink->sample_rate = ctx->inputs[0]->sample_rate; >>> + outlink->time_base = ctx->inputs[0]->time_base; >>> + outlink->channel_layout = ctx->inputs[0]->channel_layout; >>> + outlink->channels = ctx->inputs[0]->channels; >>> + >>> + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, >>> ctx->inputs[0]->channels, 1024); >>> + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, >>> ctx->inputs[1]->channels, 1024); >>> + if (!s->fifo[0] || !s->fifo[1]) >>> + return AVERROR(ENOMEM); >>> + >>> + s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); >>> + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); >>> + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); >>> + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); >>> + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); >>> + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) >>> + return AVERROR(ENOMEM); >>> + >>> + s->nb_channels = outlink->channels; >>> + s->nb_coef_channels = ctx->inputs[1]->channels; >>> + s->pts = AV_NOPTS_VALUE; >>> + >>> + return 0; >>> +} >>> + >>> +static av_cold void uninit(AVFilterContext *ctx) >>> +{ >>> + FIRContext *s = ctx->priv; >>> + int ch; >>> + >>> + if (s->sum) { >>> + for (ch = 0; ch < s->nb_channels; ch++) { >>> + av_freep(&s->sum[ch]); >>> + } >>> + } >>> + av_freep(&s->sum); >>> + >>> + if (s->coeff) { >>> + for (ch = 0; ch < s->nb_coef_channels; ch++) { >>> + av_freep(&s->coeff[ch]); >>> + } >>> + } >>> + av_freep(&s->coeff); >>> + >>> + if (s->block) { >>> + for (ch = 0; ch < s->nb_channels; ch++) { >>> + av_freep(&s->block[ch]); >>> + } >>> + } >>> + av_freep(&s->block); >>> + >>> + if (s->rdft) { >>> + for (ch = 0; ch < s->nb_channels; ch++) { >>> + av_rdft_end(s->rdft[ch]); >>> + } >>> + } >>> + av_freep(&s->rdft); >>> + >>> + if (s->irdft) { >>> + for (ch = 0; ch < s->nb_channels; ch++) { >>> + av_rdft_end(s->irdft[ch]); >>> + } >>> + } >>> + av_freep(&s->irdft); >>> + >>> + av_frame_free(&s->in[0]); >>> + av_frame_free(&s->in[1]); >>> + av_frame_free(&s->buffer); >>> + >>> + av_audio_fifo_free(s->fifo[0]); >>> + av_audio_fifo_free(s->fifo[1]); >>> +} >>> + >>> +static const AVFilterPad afir_inputs[] = { >>> + { >>> + .name = "main", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + .filter_frame = filter_frame, >>> + },{ >>> + .name = "ir", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + .filter_frame = read_ir, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +static const AVFilterPad afir_outputs[] = { >>> + { >>> + .name = "default", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + .config_props = config_output, >>> + .request_frame = request_frame, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >>> +#define OFFSET(x) offsetof(FIRContext, x) >>> + >>> +static const AVOption afir_options[] = { >>> + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, >>> {.dbl=1}, 0, 1, AF }, >>> + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, >>> {.dbl=1}, 0, 1, AF }, >>> + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, >>> {.i64=1}, 0, 1, AF }, >>> + { NULL } >>> +}; >>> + >>> +AVFILTER_DEFINE_CLASS(afir); >>> + >>> +AVFilter ff_af_afir = { >>> + .name = "afir", >>> + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response >>> filter with supplied coefficients in 2nd stream."), >>> + .priv_size = sizeof(FIRContext), >>> + .priv_class = &afir_class, >>> + .query_formats = query_formats, >>> + .uninit = uninit, >>> + .inputs = afir_inputs, >>> + .outputs = afir_outputs, >>> + .flags = AVFILTER_FLAG_SLICE_THREADS, >>> +}; >>> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >>> index 8fb87eb..555c442 100644 >>> --- a/libavfilter/allfilters.c >>> +++ b/libavfilter/allfilters.c >>> @@ -50,6 +50,7 @@ static void register_all(void) >>> REGISTER_FILTER(AEVAL, aeval, af); >>> REGISTER_FILTER(AFADE, afade, af); >>> REGISTER_FILTER(AFFTFILT, afftfilt, af); >>> + REGISTER_FILTER(AFIR, afir, af); >>> REGISTER_FILTER(AFORMAT, aformat, af); >>> REGISTER_FILTER(AGATE, agate, af); >>> REGISTER_FILTER(AINTERLEAVE, ainterleave, af); >> >> Seems that the partitioned convolution code here doesn't work. I can't >> help here. >> IMHO, you should stuck to traditional convolution code. > > Never, because non-partitioned OLA/OLS is very limited in usage, and > thus considered useless. OK.
On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda@gmail.com> wrote: > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > configure | 2 + > doc/filters.texi | 10 + > libavfilter/Makefile | 1 + > libavfilter/af_afir.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 5 files changed, 498 insertions(+) > create mode 100644 libavfilter/af_afir.c > > diff --git a/configure b/configure > index b3cb5b0..0d83c6a 100755 > --- a/configure > +++ b/configure > @@ -3078,6 +3078,8 @@ unix_protocol_select="network" > # filters > afftfilt_filter_deps="avcodec" > afftfilt_filter_select="fft" > +afir_filter_deps="avcodec" > +afir_filter_select="fft" > amovie_filter_deps="avcodec avformat" > aresample_filter_deps="swresample" > ass_filter_deps="libass" > diff --git a/doc/filters.texi b/doc/filters.texi > index 119e747..ea343d1 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)" > @end example > @end itemize > > +@section afirfilter > + > +Apply an Arbitary Frequency Impulse Response filter. > + > +This filter uses second stream as FIR coefficients. > +If second stream holds single channel, it will be used > +for all input channels in first stream, otherwise > +number of channels in second stream must be same as > +number of channels in first stream. > + > @anchor{aformat} > @section aformat > > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 66c36e4..c797eb5 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o > OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o > OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o > OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o > +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o > OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o > OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o > OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o > diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c > new file mode 100644 > index 0000000..9411c9b > --- /dev/null > +++ b/libavfilter/af_afir.c > @@ -0,0 +1,484 @@ > +/* > + * Copyright (c) 2017 Paul B Mahol > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +/** > + * @file > + * An arbitrary audio FIR filter > + */ > + > +#include "libavutil/audio_fifo.h" > +#include "libavutil/common.h" > +#include "libavutil/opt.h" > +#include "libavcodec/avfft.h" > + > +#include "audio.h" > +#include "avfilter.h" > +#include "formats.h" > +#include "internal.h" > + > +#define MAX_IR_DURATION 20 > + > +typedef struct FIRContext { > + const AVClass *class; > + > + float wet_gain; > + float dry_gain; > + int auto_gain; > + > + float gain; > + > + int eof_coeffs; > + int have_coeffs; > + int nb_coeffs; > + int nb_taps; > + int part_size; > + int nb_partitions; > + int fft_length; > + int nb_channels; > + int nb_coef_channels; > + int one2many; > + int nb_samples; > + > + RDFTContext **rdft, **irdft; > + float **sum; > + float **block; > + FFTComplex **coeff; > + > + AVAudioFifo *fifo[2]; > + AVFrame *in[2]; > + AVFrame *buffer; > + int64_t pts; > +} FIRContext; > + > +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) > +{ > + FIRContext *s = ctx->priv; > + AVFrame *out = arg; > + const FFTComplex *coeff = s->coeff[ch * !s->one2many]; > + const float *src = (const float *)s->in[0]->extended_data[ch]; > + float *dst = (float *)out->extended_data[ch]; > + float *buf = (float *)s->buffer->extended_data[ch]; > + float *sum = s->sum[ch]; > + float *block = s->block[ch]; > + int n, i; > + > + memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1)); > + memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1)); > + for (n = 0; n < s->nb_samples; n++) { > + block[n] = src[n] * s->dry_gain; > + } > + > + av_rdft_calc(s->rdft[ch], block); > + block[s->part_size / 2] = block[1]; block[s->part_size * 2] > + block[1] = 0; > + > + for (i = 0; i < s->nb_partitions; i++) { > + const int coffset = i * (s->part_size + 1); > + > + for (n = 0; n <= s->part_size; n++) { > + const float re = block[2 * n ]; > + const float im = block[2 * n + 1]; > + const float cre = coeff[coffset + n].re; > + const float cim = coeff[coffset + n].im; > + > + sum[2 * n ] += re * cre - im * cim; > + sum[2 * n + 1] += re * cim + im * cre; > + } > + } > + > + sum[1] = sum[n]; sum[1] = sum[s->part_size * 2]; > + av_rdft_calc(s->irdft[ch], sum); > + > + for (n = 0; n < out->nb_samples; n++) { > + float sample; > + > + sample = sum[out->nb_samples + n]; > + dst[n] = sample * s->wet_gain * s->gain; > + buf[n] = sum[n]; > + } > + > + return 0; > +} > + > +static int fir_frame(FIRContext *s, AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + AVFrame *out; > + > + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0])); > + > + out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? s->nb_samples : s->part_size / 2); > + if (!out) > + return AVERROR(ENOMEM); > + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples); > + if (!s->in[0]) { > + av_frame_free(&out); > + return AVERROR(ENOMEM); > + } > + > + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples); > + > + ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); > + > + av_audio_fifo_drain(s->fifo[0], out->nb_samples); > + > + out->pts = s->pts; > + if (s->pts != AV_NOPTS_VALUE) > + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); > + > + av_frame_free(&s->in[0]); > + > + return ff_filter_frame(outlink, out); > +} > + > +static int convert_coeffs(AVFilterContext *ctx) > +{ > + FIRContext *s = ctx->priv; > + int max_nb_taps, i, ch, n, N; > + float power = 0; > + > + s->nb_taps = av_audio_fifo_size(s->fifo[1]); > + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; > + if (s->nb_taps > max_nb_taps) { > + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", s->nb_taps, max_nb_taps); > + return AVERROR(EINVAL); > + } > + > + for (n = 1; (1 << n) < s->nb_taps; n++); > + N = FFMIN(n, 16); > + s->fft_length = 1 << n; > + s->part_size = 1 << (N - 1); > + s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size; > + s->nb_coeffs = s->fft_length + s->nb_partitions; > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->sum[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->sum)); > + if (!s->sum[ch]) > + return AVERROR(ENOMEM); > + } > + > + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { > + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff)); > + if (!s->coeff[ch]) > + return AVERROR(ENOMEM); > + } > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->block[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->block)); > + if (!s->block[ch]) > + return AVERROR(ENOMEM); > + } > + > + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); > + if (!s->buffer) > + return AVERROR(ENOMEM); > + > + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { > + s->rdft[ch] = av_rdft_init(N, DFT_R2C); > + s->irdft[ch] = av_rdft_init(N, IDFT_C2R); > + if (!s->rdft[ch] || !s->irdft[ch]) > + return AVERROR(ENOMEM); > + } > + > + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps); > + if (!s->in[1]) > + return AVERROR(ENOMEM); > + > + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps); > + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { > + const float *re = (const float *)s->in[1]->extended_data[!s->one2many * ch]; > + float *block = s->block[ch]; > + FFTComplex *coeff = s->coeff[ch]; > + > + for (i = 0; i < s->nb_partitions; i++) { > + const int offset = i * s->part_size; > + const int coffset = i * (s->part_size + 1); > + const int remaining = s->nb_taps - (i * s->part_size); > + const int size = remaining >= s->part_size ? s->part_size : remaining; > + > + memset(block, 0, sizeof(*block) * (2 * (s->part_size + 1))); > + for (n = 0; n < size; n++) { > + block[n] = re[n + offset]; > + power += block[n] * block[n]; > + } > + > + av_rdft_calc(s->rdft[0], block); > + > + coeff[coffset].re = block[0]; > + coeff[coffset].im = 0; > + for (n = 1; n < s->part_size; n++) { > + coeff[coffset + n].re = block[2 * n]; > + coeff[coffset + n].im = block[2 * n + 1]; > + } > + coeff[coffset + n].re = block[1]; > + coeff[coffset + n].im = 0; > + } > + } > + power /= ctx->inputs[1]->channels; > + > + av_frame_free(&s->in[1]); > + s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : sqrtf(s->part_size)); > + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N); > + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); > + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); > + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions); > + > + s->have_coeffs = 1; > + > + return 0; > +} > + > +static int read_ir(AVFilterLink *link, AVFrame *frame) > +{ > + AVFilterContext *ctx = link->dst; > + FIRContext *s = ctx->priv; > + int nb_taps, max_nb_taps; > + > + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data, > + frame->nb_samples); > + av_frame_free(&frame); > + > + nb_taps = av_audio_fifo_size(s->fifo[1]); > + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; > + if (s->nb_taps > max_nb_taps) { > + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps); > + return AVERROR(EINVAL); > + } > + > + return 0; > +} > + > +static int filter_frame(AVFilterLink *link, AVFrame *frame) > +{ > + AVFilterContext *ctx = link->dst; > + FIRContext *s = ctx->priv; > + AVFilterLink *outlink = ctx->outputs[0]; > + int ret = 0; > + > + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data, > + frame->nb_samples); > + if (s->pts == AV_NOPTS_VALUE) > + s->pts = frame->pts; > + > + av_frame_free(&frame); > + > + if (!s->have_coeffs && s->eof_coeffs) { > + ret = convert_coeffs(ctx); > + if (ret < 0) > + return ret; > + } > + > + if (s->have_coeffs) { > + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) { > + ret = fir_frame(s, outlink); > + if (ret < 0) > + break; > + } > + } > + return ret; > +} > + > +static int request_frame(AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + FIRContext *s = ctx->priv; > + int ret; > + > + if (!s->eof_coeffs) { > + ret = ff_request_frame(ctx->inputs[1]); > + if (ret == AVERROR_EOF) { > + s->eof_coeffs = 1; > + ret = 0; > + } > + return ret; > + } > + ret = ff_request_frame(ctx->inputs[0]); > + if (ret == AVERROR_EOF && s->have_coeffs) { > + while (av_audio_fifo_size(s->fifo[0]) > 0) { > + ret = fir_frame(s, outlink); > + if (ret < 0) > + return ret; > + } > + ret = AVERROR_EOF; > + } > + return ret; > +} > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats; > + AVFilterChannelLayouts *layouts = NULL; > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_FLTP, > + AV_SAMPLE_FMT_NONE > + }; > + int ret, i; > + > + layouts = ff_all_channel_counts(); > + if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0) > + return ret; > + > + for (i = 0; i < 2; i++) { > + layouts = ff_all_channel_counts(); > + if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0) > + return ret; > + } > + > + formats = ff_make_format_list(sample_fmts); > + if ((ret = ff_set_common_formats(ctx, formats)) < 0) > + return ret; > + > + formats = ff_all_samplerates(); > + return ff_set_common_samplerates(ctx, formats); > +} > + > +static int config_output(AVFilterLink *outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + FIRContext *s = ctx->priv; > + > + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels && > + ctx->inputs[1]->channels != 1) { > + av_log(ctx, AV_LOG_ERROR, > + "Second input must have same number of channels as first input or " > + "exactly 1 channel.\n"); > + return AVERROR(EINVAL); > + } > + > + s->one2many = ctx->inputs[1]->channels == 1; > + outlink->sample_rate = ctx->inputs[0]->sample_rate; > + outlink->time_base = ctx->inputs[0]->time_base; > + outlink->channel_layout = ctx->inputs[0]->channel_layout; > + outlink->channels = ctx->inputs[0]->channels; > + > + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024); > + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024); > + if (!s->fifo[0] || !s->fifo[1]) > + return AVERROR(ENOMEM); > + > + s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); > + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); > + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); > + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); > + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); > + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) > + return AVERROR(ENOMEM); > + > + s->nb_channels = outlink->channels; > + s->nb_coef_channels = ctx->inputs[1]->channels; > + s->pts = AV_NOPTS_VALUE; > + > + return 0; > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + FIRContext *s = ctx->priv; > + int ch; > + > + if (s->sum) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_freep(&s->sum[ch]); > + } > + } > + av_freep(&s->sum); > + > + if (s->coeff) { > + for (ch = 0; ch < s->nb_coef_channels; ch++) { > + av_freep(&s->coeff[ch]); > + } > + } > + av_freep(&s->coeff); > + > + if (s->block) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_freep(&s->block[ch]); > + } > + } > + av_freep(&s->block); > + > + if (s->rdft) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_rdft_end(s->rdft[ch]); > + } > + } > + av_freep(&s->rdft); > + > + if (s->irdft) { > + for (ch = 0; ch < s->nb_channels; ch++) { > + av_rdft_end(s->irdft[ch]); > + } > + } > + av_freep(&s->irdft); > + > + av_frame_free(&s->in[0]); > + av_frame_free(&s->in[1]); > + av_frame_free(&s->buffer); > + > + av_audio_fifo_free(s->fifo[0]); > + av_audio_fifo_free(s->fifo[1]); > +} > + > +static const AVFilterPad afir_inputs[] = { > + { > + .name = "main", > + .type = AVMEDIA_TYPE_AUDIO, > + .filter_frame = filter_frame, > + },{ > + .name = "ir", > + .type = AVMEDIA_TYPE_AUDIO, > + .filter_frame = read_ir, > + }, > + { NULL } > +}; > + > +static const AVFilterPad afir_outputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + .config_props = config_output, > + .request_frame = request_frame, > + }, > + { NULL } > +}; > + > +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM > +#define OFFSET(x) offsetof(FIRContext, x) > + > +static const AVOption afir_options[] = { > + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, > + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, > + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, > + { NULL } > +}; > + > +AVFILTER_DEFINE_CLASS(afir); > + > +AVFilter ff_af_afir = { > + .name = "afir", > + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."), > + .priv_size = sizeof(FIRContext), > + .priv_class = &afir_class, > + .query_formats = query_formats, > + .uninit = uninit, > + .inputs = afir_inputs, > + .outputs = afir_outputs, > + .flags = AVFILTER_FLAG_SLICE_THREADS, > +}; > diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c > index 8fb87eb..555c442 100644 > --- a/libavfilter/allfilters.c > +++ b/libavfilter/allfilters.c > @@ -50,6 +50,7 @@ static void register_all(void) > REGISTER_FILTER(AEVAL, aeval, af); > REGISTER_FILTER(AFADE, afade, af); > REGISTER_FILTER(AFFTFILT, afftfilt, af); > + REGISTER_FILTER(AFIR, afir, af); > REGISTER_FILTER(AFORMAT, aformat, af); > REGISTER_FILTER(AGATE, agate, af); > REGISTER_FILTER(AINTERLEAVE, ainterleave, af); > -- > 2.9.3 > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
diff --git a/configure b/configure index b3cb5b0..0d83c6a 100755 --- a/configure +++ b/configure @@ -3078,6 +3078,8 @@ unix_protocol_select="network" # filters afftfilt_filter_deps="avcodec" afftfilt_filter_select="fft" +afir_filter_deps="avcodec" +afir_filter_select="fft" amovie_filter_deps="avcodec avformat" aresample_filter_deps="swresample" ass_filter_deps="libass" diff --git a/doc/filters.texi b/doc/filters.texi index 119e747..ea343d1 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -878,6 +878,16 @@ afftfilt="1-clip((b/nb)*b,0,1)" @end example @end itemize +@section afirfilter + +Apply an Arbitary Frequency Impulse Response filter. + +This filter uses second stream as FIR coefficients. +If second stream holds single channel, it will be used +for all input channels in first stream, otherwise +number of channels in second stream must be same as +number of channels in first stream. + @anchor{aformat} @section aformat diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 66c36e4..c797eb5 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c new file mode 100644 index 0000000..9411c9b --- /dev/null +++ b/libavfilter/af_afir.c @@ -0,0 +1,484 @@ +/* + * Copyright (c) 2017 Paul B Mahol + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * An arbitrary audio FIR filter + */ + +#include "libavutil/audio_fifo.h" +#include "libavutil/common.h" +#include "libavutil/opt.h" +#include "libavcodec/avfft.h" + +#include "audio.h" +#include "avfilter.h" +#include "formats.h" +#include "internal.h" + +#define MAX_IR_DURATION 20 + +typedef struct FIRContext { + const AVClass *class; + + float wet_gain; + float dry_gain; + int auto_gain; + + float gain; + + int eof_coeffs; + int have_coeffs; + int nb_coeffs; + int nb_taps; + int part_size; + int nb_partitions; + int fft_length; + int nb_channels; + int nb_coef_channels; + int one2many; + int nb_samples; + + RDFTContext **rdft, **irdft; + float **sum; + float **block; + FFTComplex **coeff; + + AVAudioFifo *fifo[2]; + AVFrame *in[2]; + AVFrame *buffer; + int64_t pts; +} FIRContext; + +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +{ + FIRContext *s = ctx->priv; + AVFrame *out = arg; + const FFTComplex *coeff = s->coeff[ch * !s->one2many]; + const float *src = (const float *)s->in[0]->extended_data[ch]; + float *dst = (float *)out->extended_data[ch]; + float *buf = (float *)s->buffer->extended_data[ch]; + float *sum = s->sum[ch]; + float *block = s->block[ch]; + int n, i; + + memset(sum, 0, sizeof(*sum) * 2 * (s->part_size + 1)); + memset(block, 0, sizeof(*block) * 2 * (s->part_size + 1)); + for (n = 0; n < s->nb_samples; n++) { + block[n] = src[n] * s->dry_gain; + } + + av_rdft_calc(s->rdft[ch], block); + block[s->part_size / 2] = block[1]; + block[1] = 0; + + for (i = 0; i < s->nb_partitions; i++) { + const int coffset = i * (s->part_size + 1); + + for (n = 0; n <= s->part_size; n++) { + const float re = block[2 * n ]; + const float im = block[2 * n + 1]; + const float cre = coeff[coffset + n].re; + const float cim = coeff[coffset + n].im; + + sum[2 * n ] += re * cre - im * cim; + sum[2 * n + 1] += re * cim + im * cre; + } + } + + sum[1] = sum[n]; + av_rdft_calc(s->irdft[ch], sum); + + for (n = 0; n < out->nb_samples; n++) { + float sample; + + sample = sum[out->nb_samples + n]; + dst[n] = sample * s->wet_gain * s->gain; + buf[n] = sum[n]; + } + + return 0; +} + +static int fir_frame(FIRContext *s, AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AVFrame *out; + + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0])); + + out = ff_get_audio_buffer(outlink, s->nb_samples < s->part_size / 2 ? s->nb_samples : s->part_size / 2); + if (!out) + return AVERROR(ENOMEM); + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples); + if (!s->in[0]) { + av_frame_free(&out); + return AVERROR(ENOMEM); + } + + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples); + + ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels); + + av_audio_fifo_drain(s->fifo[0], out->nb_samples); + + out->pts = s->pts; + if (s->pts != AV_NOPTS_VALUE) + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); + + av_frame_free(&s->in[0]); + + return ff_filter_frame(outlink, out); +} + +static int convert_coeffs(AVFilterContext *ctx) +{ + FIRContext *s = ctx->priv; + int max_nb_taps, i, ch, n, N; + float power = 0; + + s->nb_taps = av_audio_fifo_size(s->fifo[1]); + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; + if (s->nb_taps > max_nb_taps) { + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", s->nb_taps, max_nb_taps); + return AVERROR(EINVAL); + } + + for (n = 1; (1 << n) < s->nb_taps; n++); + N = FFMIN(n, 16); + s->fft_length = 1 << n; + s->part_size = 1 << (N - 1); + s->nb_partitions = (s->fft_length + s->part_size - 1) / s->part_size; + s->nb_coeffs = s->fft_length + s->nb_partitions; + + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { + s->sum[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->sum)); + if (!s->sum[ch]) + return AVERROR(ENOMEM); + } + + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff)); + if (!s->coeff[ch]) + return AVERROR(ENOMEM); + } + + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { + s->block[ch] = av_calloc(2 * (s->part_size + 1), sizeof(**s->block)); + if (!s->block[ch]) + return AVERROR(ENOMEM); + } + + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size); + if (!s->buffer) + return AVERROR(ENOMEM); + + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) { + s->rdft[ch] = av_rdft_init(N, DFT_R2C); + s->irdft[ch] = av_rdft_init(N, IDFT_C2R); + if (!s->rdft[ch] || !s->irdft[ch]) + return AVERROR(ENOMEM); + } + + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps); + if (!s->in[1]) + return AVERROR(ENOMEM); + + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps); + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { + const float *re = (const float *)s->in[1]->extended_data[!s->one2many * ch]; + float *block = s->block[ch]; + FFTComplex *coeff = s->coeff[ch]; + + for (i = 0; i < s->nb_partitions; i++) { + const int offset = i * s->part_size; + const int coffset = i * (s->part_size + 1); + const int remaining = s->nb_taps - (i * s->part_size); + const int size = remaining >= s->part_size ? s->part_size : remaining; + + memset(block, 0, sizeof(*block) * (2 * (s->part_size + 1))); + for (n = 0; n < size; n++) { + block[n] = re[n + offset]; + power += block[n] * block[n]; + } + + av_rdft_calc(s->rdft[0], block); + + coeff[coffset].re = block[0]; + coeff[coffset].im = 0; + for (n = 1; n < s->part_size; n++) { + coeff[coffset + n].re = block[2 * n]; + coeff[coffset + n].im = block[2 * n + 1]; + } + coeff[coffset + n].re = block[1]; + coeff[coffset + n].im = 0; + } + } + power /= ctx->inputs[1]->channels; + + av_frame_free(&s->in[1]); + s->gain = (1.f / (1 << N)) / (s->auto_gain ? sqrtf(power) : sqrtf(s->part_size)); + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", 1 << N); + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); + av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->fft_length); + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions); + + s->have_coeffs = 1; + + return 0; +} + +static int read_ir(AVFilterLink *link, AVFrame *frame) +{ + AVFilterContext *ctx = link->dst; + FIRContext *s = ctx->priv; + int nb_taps, max_nb_taps; + + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data, + frame->nb_samples); + av_frame_free(&frame); + + nb_taps = av_audio_fifo_size(s->fifo[1]); + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate; + if (s->nb_taps > max_nb_taps) { + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps); + return AVERROR(EINVAL); + } + + return 0; +} + +static int filter_frame(AVFilterLink *link, AVFrame *frame) +{ + AVFilterContext *ctx = link->dst; + FIRContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + int ret = 0; + + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data, + frame->nb_samples); + if (s->pts == AV_NOPTS_VALUE) + s->pts = frame->pts; + + av_frame_free(&frame); + + if (!s->have_coeffs && s->eof_coeffs) { + ret = convert_coeffs(ctx); + if (ret < 0) + return ret; + } + + if (s->have_coeffs) { + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) { + ret = fir_frame(s, outlink); + if (ret < 0) + break; + } + } + return ret; +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + FIRContext *s = ctx->priv; + int ret; + + if (!s->eof_coeffs) { + ret = ff_request_frame(ctx->inputs[1]); + if (ret == AVERROR_EOF) { + s->eof_coeffs = 1; + ret = 0; + } + return ret; + } + ret = ff_request_frame(ctx->inputs[0]); + if (ret == AVERROR_EOF && s->have_coeffs) { + while (av_audio_fifo_size(s->fifo[0]) > 0) { + ret = fir_frame(s, outlink); + if (ret < 0) + return ret; + } + ret = AVERROR_EOF; + } + return ret; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE + }; + int ret, i; + + layouts = ff_all_channel_counts(); + if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0) + return ret; + + for (i = 0; i < 2; i++) { + layouts = ff_all_channel_counts(); + if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0) + return ret; + } + + formats = ff_make_format_list(sample_fmts); + if ((ret = ff_set_common_formats(ctx, formats)) < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + FIRContext *s = ctx->priv; + + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels && + ctx->inputs[1]->channels != 1) { + av_log(ctx, AV_LOG_ERROR, + "Second input must have same number of channels as first input or " + "exactly 1 channel.\n"); + return AVERROR(EINVAL); + } + + s->one2many = ctx->inputs[1]->channels == 1; + outlink->sample_rate = ctx->inputs[0]->sample_rate; + outlink->time_base = ctx->inputs[0]->time_base; + outlink->channel_layout = ctx->inputs[0]->channel_layout; + outlink->channels = ctx->inputs[0]->channels; + + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024); + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024); + if (!s->fifo[0] || !s->fifo[1]) + return AVERROR(ENOMEM); + + s->sum = av_calloc(outlink->channels, sizeof(*s->sum)); + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff)); + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block)); + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft)); + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft)); + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft) + return AVERROR(ENOMEM); + + s->nb_channels = outlink->channels; + s->nb_coef_channels = ctx->inputs[1]->channels; + s->pts = AV_NOPTS_VALUE; + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + FIRContext *s = ctx->priv; + int ch; + + if (s->sum) { + for (ch = 0; ch < s->nb_channels; ch++) { + av_freep(&s->sum[ch]); + } + } + av_freep(&s->sum); + + if (s->coeff) { + for (ch = 0; ch < s->nb_coef_channels; ch++) { + av_freep(&s->coeff[ch]); + } + } + av_freep(&s->coeff); + + if (s->block) { + for (ch = 0; ch < s->nb_channels; ch++) { + av_freep(&s->block[ch]); + } + } + av_freep(&s->block); + + if (s->rdft) { + for (ch = 0; ch < s->nb_channels; ch++) { + av_rdft_end(s->rdft[ch]); + } + } + av_freep(&s->rdft); + + if (s->irdft) { + for (ch = 0; ch < s->nb_channels; ch++) { + av_rdft_end(s->irdft[ch]); + } + } + av_freep(&s->irdft); + + av_frame_free(&s->in[0]); + av_frame_free(&s->in[1]); + av_frame_free(&s->buffer); + + av_audio_fifo_free(s->fifo[0]); + av_audio_fifo_free(s->fifo[1]); +} + +static const AVFilterPad afir_inputs[] = { + { + .name = "main", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + },{ + .name = "ir", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = read_ir, + }, + { NULL } +}; + +static const AVFilterPad afir_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + .request_frame = request_frame, + }, + { NULL } +}; + +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM +#define OFFSET(x) offsetof(FIRContext, x) + +static const AVOption afir_options[] = { + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(afir); + +AVFilter ff_af_afir = { + .name = "afir", + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."), + .priv_size = sizeof(FIRContext), + .priv_class = &afir_class, + .query_formats = query_formats, + .uninit = uninit, + .inputs = afir_inputs, + .outputs = afir_outputs, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 8fb87eb..555c442 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -50,6 +50,7 @@ static void register_all(void) REGISTER_FILTER(AEVAL, aeval, af); REGISTER_FILTER(AFADE, afade, af); REGISTER_FILTER(AFFTFILT, afftfilt, af); + REGISTER_FILTER(AFIR, afir, af); REGISTER_FILTER(AFORMAT, aformat, af); REGISTER_FILTER(AGATE, agate, af); REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- configure | 2 + doc/filters.texi | 10 + libavfilter/Makefile | 1 + libavfilter/af_afir.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 5 files changed, 498 insertions(+) create mode 100644 libavfilter/af_afir.c