[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Paul B Mahol
onemda at gmail.com
Fri May 5 22:30:38 EEST 2017
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
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);
--
2.9.3
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