On Fri, 14 Feb 2014 02:56:46 -0500, Andrew Kelley <superjo...@gmail.com> wrote:
> This patch adds the `compand` audio filter from ffmpeg master branch
> (currently at 7f0f47b3df) adapted to work with libav.
>
> The following changes are made:
>
> * use float instead of double
> * use strtok_r instead of av_strtok
> * remove asserts
> * use AVFrame instead of manually allocating memory
> ---
> Changelog | 1 +
> doc/filters.texi | 74 +++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_compand.c | 548
> +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> libavfilter/version.h | 2 +-
> 6 files changed, 626 insertions(+), 1 deletion(-)
> create mode 100644 libavfilter/af_compand.c
>
> diff --git a/Changelog b/Changelog
> index bed6c31..74b7f1a 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -58,6 +58,7 @@ version 10:
> - Mirillis FIC video decoder
> - Support DNx444
> - libx265 encoder
> +- compand audio filter
>
>
> version 9:
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 8c83b4e..8863384 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -467,6 +467,80 @@ To fix a 5.1 WAV improperly encoded in AAC's native
> channel order
> avconv -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav
> @end example
>
> +@section compand
> +Compress or expand audio dynamic range.
> +
> +A description of the accepted options follows.
> +
> +@table @option
> +
> +@item attacks
> +@item decays
> +Set list of times in seconds for each channel over which the instantaneous
> level
> +of the input signal is averaged to determine its volume. @var{attacks}
> refers to
> +increase of volume and @var{decays} refers to decrease of volume. For most
> +situations, the attack time (response to the audio getting louder) should be
> +shorter than the decay time because the human ear is more sensitive to sudden
> +loud audio than sudden soft audio. Typical value for attack is 0.3 seconds
> and
> +for decay 0.8 seconds.
> +
> +@item points
> +Set list of points for transfer function, specified in dB relative to maximum
> +possible signal amplitude. Each key points list need to be defined using the
> +following syntax: @code{x0/y0 x1/y1 x2/y2 ....}
> +
> +The input values must be in strictly increasing order but the transfer
> function
> +does not have to be monotonically rising. The point @code{0/0} is assumed but
> +may be overridden (by @code{0/out-dBn}). Typical values for the transfer
> +function are @code{-70/-70 -60/-20}.
> +
> +@item soft-knee
> +Set amount for which the points at where adjacent line segments on the
> transfer
> +function meet will be rounded. Defaults is 0.01.
> +
> +@item gain
> +Set additional gain in dB to be applied at all points on the transfer
> function
> +and allows easy adjustment of the overall gain. Default is 0.
> +
> +@item volume
> +Set initial volume in dB to be assumed for each channel when filtering
> starts.
> +This permits the user to supply a nominal level initially, so that, for
> +example, a very large gain is not applied to initial signal levels before the
> +companding has begun to operate. A typical value for audio which is initially
> +quiet is -90 dB. Default is 0.
> +
> +@item delay
> +Set delay in seconds. Default is 0. The input audio is analysed immediately,
> +but audio is delayed before being fed to the volume adjuster. Specifying a
> +delay approximately equal to the attack/decay times allows the filter to
> +effectively operate in predictive rather than reactive mode.
> +
> +@end table
> +
> +@subsection Examples
> +
> +@itemize
> +@item
> +Make music with both quiet and loud passages suitable for listening in a
> noisy
> +environment:
> +@example
> +compand=.3 .3:1 1:-90/-60 -60/-40 -40/-30 -20/-20:6:0:-90:0.2
I see the elements in the list are divided by spaces, which doesn't seem to be
mentioned anywhere. Also, other filters usually use '|' for this.
> +@end example
> +
> +@item
> +Noise-gate for when the noise is at a lower level than the signal:
> +@example
> +compand=.1 .1:.2 .2:-900/-900 -50.1/-900 -50/-50:.01:0:-90:.1
> +@end example
> +
> +@item
> +Here is another noise-gate, this time for when the noise is at a higher level
> +than the signal (making it, in some ways, similar to squelch):
> +@example
> +compand=.1 .1:.1 .1:-45.1/-45.1 -45/-900 0/-900:.01:45:-90:.1
> +@end example
> +@end itemize
> +
> @section join
> Join multiple input streams into one multi-channel stream.
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 92c1561..2badb3e 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -37,6 +37,7 @@ OBJS-$(CONFIG_CHANNELSPLIT_FILTER) +=
> af_channelsplit.o
> OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
> OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
> OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
> +OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
>
> OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
>
> diff --git a/libavfilter/af_compand.c b/libavfilter/af_compand.c
> new file mode 100644
> index 0000000..d46dfa4
> --- /dev/null
> +++ b/libavfilter/af_compand.c
> @@ -0,0 +1,548 @@
> +/*
> + * Copyright (c) 1999 Chris Bagwell
> + * Copyright (c) 1999 Nick Bailey
> + * Copyright (c) 2007 Rob Sykes <r...@users.sourceforge.net>
> + * Copyright (c) 2013 Paul B Mahol
> + * Copyright (c) 2014 Andrew Kelley
> + *
> + * This file is part of libav.
> + *
> + * Libav 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.
> + *
> + * Libav 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 Libav; if not, write to the Free Software
> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301
> USA
> + *
> + */
> +
> +/**
> + * @file
> + * audio compand filter
> + */
> +
> +#include <string.h>
> +
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/common.h"
> +#include "libavutil/mathematics.h"
> +#include "libavutil/mem.h"
> +#include "libavutil/opt.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +typedef struct ChanParam {
> + float attack;
> + float decay;
> + float volume;
> +} ChanParam;
> +
> +typedef struct CompandSegment {
> + float x, y;
> + float a, b;
> +} CompandSegment;
> +
> +typedef struct CompandContext {
> + const AVClass *class;
> + int nb_channels;
> + char *attacks, *decays, *points;
> + CompandSegment *segments;
> + ChanParam *channels;
> + float in_min_lin;
> + float out_min_lin;
> + float curve_dB;
> + float gain_dB;
> + float initial_volume;
> + float delay;
> + AVFrame *delay_frame;
> + int delay_samples;
> + int delay_count;
> + int delay_index;
> + int64_t pts;
> +
> + int (*compand)(AVFilterContext *ctx, AVFrame *frame);
> +} CompandContext;
> +
> +#define OFFSET(x) offsetof(CompandContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM
> +
> +static const AVOption compand_options[] = {
> + { "attacks", "set time over which increase of volume is determined",
> OFFSET(attacks), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "decays", "set time over which decrease of volume is determined",
> OFFSET(decays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "points", "set points of transfer function", OFFSET(points),
> AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_FLOAT,
> {.dbl=0.01}, 0.01, 900, A },
> + { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_FLOAT,
> {.dbl=0}, -900, 900, A },
> + { "volume", "set initial volume", OFFSET(initial_volume),
> AV_OPT_TYPE_FLOAT, {.dbl=0}, -900, 0, A },
> + { "delay", "set delay for samples before sending them to volume
> adjuster", OFFSET(delay), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 20, A },
> + { NULL }
> +};
> +
> +static const AVClass compand_class = {
> + .class_name = "compand filter",
> + .item_name = av_default_item_name,
> + .option = compand_options,
> + .version = LIBAVUTIL_VERSION_INT,
> +};
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> + CompandContext *s = ctx->priv;
> +
> + if (!s->attacks || !s->decays || !s->points) {
> + av_log(ctx, AV_LOG_ERROR, "Missing attacks and/or decays and/or
> points.\n");
> + return AVERROR(EINVAL);
> + }
> +
> + return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + CompandContext *s = ctx->priv;
> +
> + av_freep(&s->channels);
> + av_freep(&s->segments);
> + av_frame_free(&s->delay_frame);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterChannelLayouts *layouts;
> + AVFilterFormats *formats;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_FLTP,
> + AV_SAMPLE_FMT_NONE
> + };
> +
> + layouts = ff_all_channel_layouts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ff_set_common_channel_layouts(ctx, layouts);
> +
> + formats = ff_make_format_list(sample_fmts);
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_formats(ctx, formats);
> +
> + formats = ff_all_samplerates();
> + if (!formats)
> + return AVERROR(ENOMEM);
> + ff_set_common_samplerates(ctx, formats);
> +
> + return 0;
> +}
> +
> +static void count_items(char *item_str, int *nb_items)
> +{
> + char *p;
> +
> + *nb_items = 1;
> + for (p = item_str; *p; p++) {
> + if (*p == ' ')
> + (*nb_items)++;
> + }
> +
> +}
> +
> +static void update_volume(ChanParam *cp, float in)
> +{
> + float delta = in - cp->volume;
> +
> + if (delta > 0.0)
> + cp->volume += delta * cp->attack;
> + else
> + cp->volume += delta * cp->decay;
> +}
> +
> +static float get_volume(CompandContext *s, float in_lin)
> +{
> + CompandSegment *cs;
> + float in_log, out_log;
> + int i;
> +
> + if (in_lin < s->in_min_lin)
> + return s->out_min_lin;
> +
> + in_log = log(in_lin);
> +
> + for (i = 1;; i++)
> + if (in_log <= s->segments[i + 1].x)
> + break;
> +
> + cs = &s->segments[i];
> + in_log -= cs->x;
> + out_log = cs->y + in_log * (cs->a * in_log + cs->b);
> +
> + return exp(out_log);
> +}
> +
> +static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
> +{
> + CompandContext *s = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> + const int channels = s->nb_channels;
> + const int nb_samples = frame->nb_samples;
> + AVFrame *out_frame;
> + int chan, i;
> +
> + if (av_frame_is_writable(frame)) {
> + out_frame = frame;
> + } else {
> + out_frame = ff_get_audio_buffer(inlink, nb_samples);
> + if (!out_frame)
> + return AVERROR(ENOMEM);
Leaking frame here.
> + av_frame_copy_props(out_frame, frame);
This should be checked for errors.
> + }
> +
> + for (chan = 0; chan < channels; chan++) {
> + const float *src = (float *)frame->extended_data[chan];
> + float *dst = (float *)out_frame->extended_data[chan];
> + ChanParam *cp = &s->channels[chan];
> +
> + for (i = 0; i < nb_samples; i++) {
> + update_volume(cp, fabs(src[i]));
> +
> + dst[i] = av_clipf(src[i] * get_volume(s, cp->volume), -1, 1);
> + }
> + }
> +
> + if (frame != out_frame)
> + av_frame_free(&frame);
> +
> + return ff_filter_frame(ctx->outputs[0], out_frame);
> +}
> +
> +#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
> +
> +static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
> +{
> + CompandContext *s = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> + const int channels = s->nb_channels;
> + const int nb_samples = frame->nb_samples;
> + int chan, i, av_uninit(dindex), oindex, av_uninit(count);
> + AVFrame *out_frame = NULL;
> +
> + for (chan = 0; chan < channels; chan++) {
> + AVFrame *delay_frame = s->delay_frame;
> + const float *src = (float *)frame->extended_data[chan];
> + float *dbuf = (float *)delay_frame->extended_data[chan];
> + ChanParam *cp = &s->channels[chan];
> + float *dst;
> +
> + count = s->delay_count;
> + dindex = s->delay_index;
> + for (i = 0, oindex = 0; i < nb_samples; i++) {
> + const float in = src[i];
> + update_volume(cp, fabs(in));
> +
> + if (count >= s->delay_samples) {
> + if (!out_frame) {
> + out_frame = ff_get_audio_buffer(inlink, nb_samples - i);
> + if (!out_frame)
> + return AVERROR(ENOMEM);
> + av_frame_copy_props(out_frame, frame);
Same as above -- leaking frame + error check needed
--
Anton Khirnov
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