[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Paul B Mahol
onemda at gmail.com
Sat May 6 11:54:28 EEST 2017
On 5/6/17, Muhammad Faiz <mfcc64 at gmail.com> wrote:
> On Sat, May 6, 2017 at 2:30 AM, Paul B Mahol <onemda at gmail.com> wrote:
>> Signed-off-by: Paul B Mahol <onemda at 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
>>
>> + at 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.
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