If it helps, here is the diff between af_compand.c in ffmpeg and
af_compand.c in libav:
--- /home/andy/dev/ffmpeg/libavfilter/af_compand.c 2014-02-02
12:07:39.789987354 -0500
+++ /home/andy/dev/libav/libavfilter/af_compand.c 2014-02-08
16:50:46.428038489 -0500
@@ -3,8 +3,9 @@
* 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 FFmpeg.
+ * This file is part of libav.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
@@ -22,12 +23,20 @@
*
*/
+/**
+ * @file
+ * audio compand filter
+ */
+
+#include "libavutil/mem.h"
#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-#include "avfilter.h"
+#include "libavutil/common.h"
#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
#include "internal.h"
typedef struct ChanParam {
@@ -62,7 +71,7 @@
} CompandContext;
#define OFFSET(x) offsetof(CompandContext, x)
-#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#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
},
@@ -75,7 +84,12 @@
{ NULL }
};
-AVFILTER_DEFINE_CLASS(compand);
+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)
{
@@ -171,11 +185,26 @@
return exp(out_log);
}
+/**
+ * Clip a double value into the amin-amax range.
+ * @param a value to clip
+ * @param amin minimum value of the clip range
+ * @param amax maximum value of the clip range
+ * @return clipped value
+ */
+static av_always_inline av_const double av_clipd_c(double a, double
amin, double amax)
+{
+ av_assert2(amin <= amax);
+ if (a < amin) return amin;
+ else if (a > amax) return amax;
+ else return a;
+}
+
static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
{
CompandContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
- const int channels = inlink->channels;
+ const int channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
const int nb_samples = frame->nb_samples;
AVFrame *out_frame;
int chan, i;
@@ -197,7 +226,7 @@
for (i = 0; i < nb_samples; i++) {
update_volume(cp, fabs(src[i]));
- dst[i] = av_clipd(src[i] * get_volume(s, cp->volume), -1, 1);
+ dst[i] = av_clipd_c(src[i] * get_volume(s, cp->volume), -1, 1);
}
}
@@ -213,12 +242,12 @@
{
CompandContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
- const int channels = inlink->channels;
+ const int channels =
av_get_channel_layout_nb_channels(inlink->channel_layout);
const int nb_samples = frame->nb_samples;
int chan, i, av_uninit(dindex), oindex, av_uninit(count);
AVFrame *out_frame = NULL;
- av_assert1(channels > 0); /* would corrupt delay_count and delay_index */
+ av_assert2(channels > 0); /* would corrupt delay_count and delay_index */
for (chan = 0; chan < channels; chan++) {
const double *src = (double *)frame->extended_data[chan];
@@ -243,7 +272,7 @@
}
dst = (double *)out_frame->extended_data[chan];
- dst[oindex++] = av_clipd(dbuf[dindex] * get_volume(s,
cp->volume), -1, 1);
+ dst[oindex++] = av_clipd_c(dbuf[dindex] *
get_volume(s, cp->volume), -1, 1);
} else {
count++;
}
@@ -264,7 +293,7 @@
{
AVFilterContext *ctx = outlink->src;
CompandContext *s = ctx->priv;
- const int channels = outlink->channels;
+ const int channels =
av_get_channel_layout_nb_channels(outlink->channel_layout);
int chan, i, dindex;
AVFrame *frame = NULL;
@@ -281,7 +310,7 @@
dindex = s->delay_index;
for (i = 0; i < frame->nb_samples; i++) {
- dst[i] = av_clipd(dbuf[dindex] * get_volume(s, cp->volume), -1, 1);
+ dst[i] = av_clipd_c(dbuf[dindex] * get_volume(s,
cp->volume), -1, 1);
dindex = MOD(dindex + 1, s->delay_samples);
}
}
@@ -291,10 +320,56 @@
return ff_filter_frame(outlink, frame);
}
+static char *av_strtok(char *s, const char *delim, char **saveptr)
+{
+ char *tok;
+
+ if (!s && !(s = *saveptr))
+ return NULL;
+
+ /* skip leading delimiters */
+ s += strspn(s, delim);
+
+ /* s now points to the first non delimiter char, or to the end of
the string */
+ if (!*s) {
+ *saveptr = NULL;
+ return NULL;
+ }
+ tok = s++;
+
+ /* skip non delimiters */
+ s += strcspn(s, delim);
+ if (*s) {
+ *s = 0;
+ *saveptr = s+1;
+ } else {
+ *saveptr = NULL;
+ }
+
+ return tok;
+}
+
+
+static int av_samples_alloc_array_and_samples(uint8_t ***audio_data,
int *linesize, int nb_channels,
+ int nb_samples, enum
AVSampleFormat sample_fmt, int align)
+{
+ int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
+
+ *audio_data = av_mallocz(nb_planes * sizeof(**audio_data));
+ if (!*audio_data)
+ return AVERROR(ENOMEM);
+ ret = av_samples_alloc(*audio_data, linesize, nb_channels,
+ nb_samples, sample_fmt, align);
+ if (ret < 0)
+ av_freep(audio_data);
+ return ret;
+}
+
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
CompandContext *s = ctx->priv;
+ const int channels =
av_get_channel_layout_nb_channels(outlink->channel_layout);
const int sample_rate = outlink->sample_rate;
double radius = s->curve_dB * M_LN10 / 20;
int nb_attacks, nb_decays, nb_points;
@@ -306,14 +381,14 @@
count_items(s->decays, &nb_decays);
count_items(s->points, &nb_points);
- if ((nb_attacks > outlink->channels) || (nb_decays > outlink->channels)) {
+ if ((nb_attacks > channels) || (nb_decays > channels)) {
av_log(ctx, AV_LOG_ERROR, "Number of attacks/decays bigger
than number of channels.\n");
return AVERROR(EINVAL);
}
uninit(ctx);
- s->channels = av_mallocz_array(outlink->channels, sizeof(*s->channels));
+ s->channels = av_mallocz_array(channels, sizeof(*s->channels));
s->segments = av_mallocz_array((nb_points + 4) * 2, sizeof(*s->segments));
if (!s->channels || !s->segments)
@@ -434,7 +509,7 @@
s->in_min_lin = exp(s->segments[1].x);
s->out_min_lin = exp(s->segments[1].y);
- for (i = 0; i < outlink->channels; i++) {
+ for (i = 0; i < channels; i++) {
ChanParam *cp = &s->channels[i];
if (cp->attack > 1.0 / sample_rate)
@@ -452,12 +527,11 @@
if (s->delay_samples > 0) {
int ret;
if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
- outlink->channels,
+ channels,
s->delay_samples,
outlink->format, 0)) < 0)
return ret;
s->compand = compand_delay;
- outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
} else {
s->compand = compand_nodelay;
}
@@ -480,7 +554,7 @@
ret = ff_request_frame(ctx->inputs[0]);
- if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)
+ if (ret == AVERROR_EOF && s->delay_count)
ret = compand_drain(outlink);
return ret;
@@ -505,6 +579,7 @@
{ NULL }
};
+
AVFilter ff_af_compand = {
.name = "compand",
.description = NULL_IF_CONFIG_SMALL("Compress or expand audio
dynamic range."),
@@ -512,7 +587,6 @@
.priv_size = sizeof(CompandContext),
.priv_class = &compand_class,
.init = init,
- .uninit = uninit,
.inputs = compand_inputs,
.outputs = compand_outputs,
};
On Sat, Feb 8, 2014 at 8:21 PM, Andrew Kelley <superjo...@gmail.com> wrote:
> This patch adds the `compand` audio filter from ffmpeg master branch
> (currently at 7f0f47b3df9d92eea0d1de94df29971b49f10777) adapted to
> work with libav.
>
> My downstream library is successfully using this filter to allow the
> user to turn up the volume gain beyond 100%:
>
> https://github.com/andrewrk/libgroove/commit/b67b17e091d17bc9eec70f89e61fde90bbee0830
>
> Some things to consider about this filter:
>
> In ffmpeg, an AVFilterLink has the concept of FF_LINK_FLAG_REQUEST_LOOP.
> "A filter must set this flag on an output link if it may return 0 in
> request_frame() without filtering a frame".
>
> Libav does not have this concept, so I removed the flag. The change
> appears to have no effect; however perhaps someone knows better than I do.
>
> The filter makes use of `av_strtok` to parse the input parameters, and
> it appears that libav has removed that function, so I inlined it into
> the filter. Same goes for `av_clipd_c`. Same for
> `av_samples_alloc_array_and_samples` except it looks like ffmpeg added
> that function recently.
> ---
> doc/filters.texi | 74 ++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_compand.c | 592
> +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 668 insertions(+)
> create mode 100644 libavfilter/af_compand.c
>
> 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
> +@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..55d75af
> --- /dev/null
> +++ b/libavfilter/af_compand.c
> @@ -0,0 +1,592 @@
> +/*
> + * 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.
> + *
> + * 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
> + * audio compand filter
> + */
> +
> +#include "libavutil/mem.h"
> +#include "libavutil/avassert.h"
> +#include "libavutil/mathematics.h"
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/opt.h"
> +#include "libavutil/common.h"
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +typedef struct ChanParam {
> + double attack;
> + double decay;
> + double volume;
> +} ChanParam;
> +
> +typedef struct CompandSegment {
> + double x, y;
> + double a, b;
> +} CompandSegment;
> +
> +typedef struct CompandContext {
> + const AVClass *class;
> + char *attacks, *decays, *points;
> + CompandSegment *segments;
> + ChanParam *channels;
> + double in_min_lin;
> + double out_min_lin;
> + double curve_dB;
> + double gain_dB;
> + double initial_volume;
> + double delay;
> + uint8_t **delayptrs;
> + 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_DOUBLE,
> {.dbl=0.01}, 0.01, 900, A },
> + { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE,
> {.dbl=0}, -900, 900, A },
> + { "volume", "set initial volume", OFFSET(initial_volume),
> AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 0, A },
> + { "delay", "set delay for samples before sending them to volume
> adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.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);
> + if (s->delayptrs)
> + av_freep(&s->delayptrs[0]);
> + av_freep(&s->delayptrs);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterChannelLayouts *layouts;
> + AVFilterFormats *formats;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_DBLP,
> + 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, double in)
> +{
> + double delta = in - cp->volume;
> +
> + if (delta > 0.0)
> + cp->volume += delta * cp->attack;
> + else
> + cp->volume += delta * cp->decay;
> +}
> +
> +static double get_volume(CompandContext *s, double in_lin)
> +{
> + CompandSegment *cs;
> + double 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);
> +}
> +
> +/**
> + * Clip a double value into the amin-amax range.
> + * @param a value to clip
> + * @param amin minimum value of the clip range
> + * @param amax maximum value of the clip range
> + * @return clipped value
> + */
> +static av_always_inline av_const double av_clipd_c(double a, double amin,
> double amax)
> +{
> + av_assert2(amin <= amax);
> + if (a < amin) return amin;
> + else if (a > amax) return amax;
> + else return a;
> +}
> +
> +static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
> +{
> + CompandContext *s = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> + const int channels =
> av_get_channel_layout_nb_channels(inlink->channel_layout);
> + 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);
> + av_frame_copy_props(out_frame, frame);
> + }
> +
> + for (chan = 0; chan < channels; chan++) {
> + const double *src = (double *)frame->extended_data[chan];
> + double *dst = (double *)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_clipd_c(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 =
> av_get_channel_layout_nb_channels(inlink->channel_layout);
> + const int nb_samples = frame->nb_samples;
> + int chan, i, av_uninit(dindex), oindex, av_uninit(count);
> + AVFrame *out_frame = NULL;
> +
> + av_assert2(channels > 0); /* would corrupt delay_count and
> delay_index */
> +
> + for (chan = 0; chan < channels; chan++) {
> + const double *src = (double *)frame->extended_data[chan];
> + double *dbuf = (double *)s->delayptrs[chan];
> + ChanParam *cp = &s->channels[chan];
> + double *dst;
> +
> + count = s->delay_count;
> + dindex = s->delay_index;
> + for (i = 0, oindex = 0; i < nb_samples; i++) {
> + const double 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);
> + out_frame->pts = s->pts;
> + s->pts += av_rescale_q(nb_samples - i,
> (AVRational){1, inlink->sample_rate}, inlink->time_base);
> + }
> +
> + dst = (double *)out_frame->extended_data[chan];
> + dst[oindex++] = av_clipd_c(dbuf[dindex] * get_volume(s,
> cp->volume), -1, 1);
> + } else {
> + count++;
> + }
> +
> + dbuf[dindex] = in;
> + dindex = MOD(dindex + 1, s->delay_samples);
> + }
> + }
> +
> + s->delay_count = count;
> + s->delay_index = dindex;
> +
> + av_frame_free(&frame);
> + return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0;
> +}
> +
> +static int compand_drain(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + CompandContext *s = ctx->priv;
> + const int channels =
> av_get_channel_layout_nb_channels(outlink->channel_layout);
> + int chan, i, dindex;
> + AVFrame *frame = NULL;
> +
> + frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
> + if (!frame)
> + return AVERROR(ENOMEM);
> + frame->pts = s->pts;
> + s->pts += av_rescale_q(frame->nb_samples, (AVRational){1,
> outlink->sample_rate}, outlink->time_base);
> +
> + for (chan = 0; chan < channels; chan++) {
> + double *dbuf = (double *)s->delayptrs[chan];
> + double *dst = (double *)frame->extended_data[chan];
> + ChanParam *cp = &s->channels[chan];
> +
> + dindex = s->delay_index;
> + for (i = 0; i < frame->nb_samples; i++) {
> + dst[i] = av_clipd_c(dbuf[dindex] * get_volume(s, cp->volume),
> -1, 1);
> + dindex = MOD(dindex + 1, s->delay_samples);
> + }
> + }
> + s->delay_count -= frame->nb_samples;
> + s->delay_index = dindex;
> +
> + return ff_filter_frame(outlink, frame);
> +}
> +
> +static char *av_strtok(char *s, const char *delim, char **saveptr)
> +{
> + char *tok;
> +
> + if (!s && !(s = *saveptr))
> + return NULL;
> +
> + /* skip leading delimiters */
> + s += strspn(s, delim);
> +
> + /* s now points to the first non delimiter char, or to the end of the
> string */
> + if (!*s) {
> + *saveptr = NULL;
> + return NULL;
> + }
> + tok = s++;
> +
> + /* skip non delimiters */
> + s += strcspn(s, delim);
> + if (*s) {
> + *s = 0;
> + *saveptr = s+1;
> + } else {
> + *saveptr = NULL;
> + }
> +
> + return tok;
> +}
> +
> +
> +static int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int
> *linesize, int nb_channels,
> + int nb_samples, enum
> AVSampleFormat sample_fmt, int align)
> +{
> + int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ?
> nb_channels : 1;
> +
> + *audio_data = av_mallocz(nb_planes * sizeof(**audio_data));
> + if (!*audio_data)
> + return AVERROR(ENOMEM);
> + ret = av_samples_alloc(*audio_data, linesize, nb_channels,
> + nb_samples, sample_fmt, align);
> + if (ret < 0)
> + av_freep(audio_data);
> + return ret;
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + CompandContext *s = ctx->priv;
> + const int channels =
> av_get_channel_layout_nb_channels(outlink->channel_layout);
> + const int sample_rate = outlink->sample_rate;
> + double radius = s->curve_dB * M_LN10 / 20;
> + int nb_attacks, nb_decays, nb_points;
> + char *p, *saveptr = NULL;
> + int new_nb_items, num;
> + int i;
> +
> + count_items(s->attacks, &nb_attacks);
> + count_items(s->decays, &nb_decays);
> + count_items(s->points, &nb_points);
> +
> + if ((nb_attacks > channels) || (nb_decays > channels)) {
> + av_log(ctx, AV_LOG_ERROR, "Number of attacks/decays bigger than
> number of channels.\n");
> + return AVERROR(EINVAL);
> + }
> +
> + uninit(ctx);
> +
> + s->channels = av_mallocz_array(channels, sizeof(*s->channels));
> + s->segments = av_mallocz_array((nb_points + 4) * 2,
> sizeof(*s->segments));
> +
> + if (!s->channels || !s->segments)
> + return AVERROR(ENOMEM);
> +
> + p = s->attacks;
> + for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
> + char *tstr = av_strtok(p, " ", &saveptr);
> + p = NULL;
> + new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;
> + if (s->channels[i].attack < 0)
> + return AVERROR(EINVAL);
> + }
> + nb_attacks = new_nb_items;
> +
> + p = s->decays;
> + for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
> + char *tstr = av_strtok(p, " ", &saveptr);
> + p = NULL;
> + new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;
> + if (s->channels[i].decay < 0)
> + return AVERROR(EINVAL);
> + }
> + nb_decays = new_nb_items;
> +
> + if (nb_attacks != nb_decays) {
> + av_log(ctx, AV_LOG_ERROR, "Number of attacks %d differs from
> number of decays %d.\n", nb_attacks, nb_decays);
> + return AVERROR(EINVAL);
> + }
> +
> +#define S(x) s->segments[2 * ((x) + 1)]
> + p = s->points;
> + for (i = 0, new_nb_items = 0; i < nb_points; i++) {
> + char *tstr = av_strtok(p, " ", &saveptr);
> + p = NULL;
> + if (sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
> + av_log(ctx, AV_LOG_ERROR, "Invalid and/or missing
> input/output value.\n");
> + return AVERROR(EINVAL);
> + }
> + if (i && S(i - 1).x > S(i).x) {
> + av_log(ctx, AV_LOG_ERROR, "Transfer function input values
> must be increasing.\n");
> + return AVERROR(EINVAL);
> + }
> + S(i).y -= S(i).x;
> + av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
> + new_nb_items++;
> + }
> + num = new_nb_items;
> +
> + /* Add 0,0 if necessary */
> + if (num == 0 || S(num - 1).x)
> + num++;
> +
> +#undef S
> +#define S(x) s->segments[2 * (x)]
> + /* Add a tail off segment at the start */
> + S(0).x = S(1).x - 2 * s->curve_dB;
> + S(0).y = S(1).y;
> + num++;
> +
> + /* Join adjacent colinear segments */
> + for (i = 2; i < num; i++) {
> + double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
> + double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
> + int j;
> +
> + if (fabs(g1 - g2))
> + continue;
> + num--;
> + for (j = --i; j < num; j++)
> + S(j) = S(j + 1);
> + }
> +
> + for (i = 0; !i || s->segments[i - 2].x; i += 2) {
> + s->segments[i].y += s->gain_dB;
> + s->segments[i].x *= M_LN10 / 20;
> + s->segments[i].y *= M_LN10 / 20;
> + }
> +
> +#define L(x) s->segments[i - (x)]
> + for (i = 4; s->segments[i - 2].x; i += 2) {
> + double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
> +
> + L(4).a = 0;
> + L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
> +
> + L(2).a = 0;
> + L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
> +
> + theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
> + len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.));
> + r = FFMIN(radius, len);
> + L(3).x = L(2).x - r * cos(theta);
> + L(3).y = L(2).y - r * sin(theta);
> +
> + theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
> + len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.));
> + r = FFMIN(radius, len / 2);
> + x = L(2).x + r * cos(theta);
> + y = L(2).y + r * sin(theta);
> +
> + cx = (L(3).x + L(2).x + x) / 3;
> + cy = (L(3).y + L(2).y + y) / 3;
> +
> + L(2).x = x;
> + L(2).y = y;
> +
> + in1 = cx - L(3).x;
> + out1 = cy - L(3).y;
> + in2 = L(2).x - L(3).x;
> + out2 = L(2).y - L(3).y;
> + L(3).a = (out2 / in2 - out1 / in1) / (in2-in1);
> + L(3).b = out1 / in1 - L(3).a * in1;
> + }
> + L(3).x = 0;
> + L(3).y = L(2).y;
> +
> + s->in_min_lin = exp(s->segments[1].x);
> + s->out_min_lin = exp(s->segments[1].y);
> +
> + for (i = 0; i < channels; i++) {
> + ChanParam *cp = &s->channels[i];
> +
> + if (cp->attack > 1.0 / sample_rate)
> + cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
> + else
> + cp->attack = 1.0;
> + if (cp->decay > 1.0 / sample_rate)
> + cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
> + else
> + cp->decay = 1.0;
> + cp->volume = pow(10.0, s->initial_volume / 20);
> + }
> +
> + s->delay_samples = s->delay * sample_rate;
> + if (s->delay_samples > 0) {
> + int ret;
> + if ((ret = av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
> + channels,
> + s->delay_samples,
> + outlink->format,
> 0)) < 0)
> + return ret;
> + s->compand = compand_delay;
> + } else {
> + s->compand = compand_nodelay;
> + }
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + CompandContext *s = ctx->priv;
> +
> + return s->compand(ctx, frame);
> +}
> +
> +static int request_frame(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + CompandContext *s = ctx->priv;
> + int ret;
> +
> + ret = ff_request_frame(ctx->inputs[0]);
> +
> + if (ret == AVERROR_EOF && s->delay_count)
> + ret = compand_drain(outlink);
> +
> + return ret;
> +}
> +
> +static const AVFilterPad compand_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad compand_outputs[] = {
> + {
> + .name = "default",
> + .request_frame = request_frame,
> + .config_props = config_output,
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + { NULL }
> +};
> +
> +
> +AVFilter ff_af_compand = {
> + .name = "compand",
> + .description = NULL_IF_CONFIG_SMALL("Compress or expand audio
> dynamic range."),
> + .query_formats = query_formats,
> + .priv_size = sizeof(CompandContext),
> + .priv_class = &compand_class,
> + .init = init,
> + .inputs = compand_inputs,
> + .outputs = compand_outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 9702a0a..e47a22e 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -54,6 +54,7 @@ void avfilter_register_all(void)
> REGISTER_FILTER(ATRIM, atrim, af);
> REGISTER_FILTER(CHANNELMAP, channelmap, af);
> REGISTER_FILTER(CHANNELSPLIT, channelsplit, af);
> + REGISTER_FILTER(COMPAND, compand, af);
> REGISTER_FILTER(JOIN, join, af);
> REGISTER_FILTER(RESAMPLE, resample, af);
> REGISTER_FILTER(VOLUME, volume, af);
> --
> 1.8.3.2
>
>
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