On 25/04/14 19:31, Alessandro Ghedini wrote:
> ---
> configure | 4 +
> doc/filters.texi | 30 +++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_bs2b.c | 213
> +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 5 files changed, 249 insertions(+)
> create mode 100644 libavfilter/af_bs2b.c
>
> diff --git a/configure b/configure
> index d088e26..a5de282 100755
> --- a/configure
> +++ b/configure
> @@ -178,6 +178,7 @@ External library support:
> --enable-bzlib enable bzlib [autodetect]
> --enable-frei0r enable frei0r video filtering
> --enable-gnutls enable gnutls [no]
> + --enable-libbs2b enable bs2b DSP library [no]
> --enable-libcdio enable audio CD grabbing with libcdio
> --enable-libdc1394 enable IIDC-1394 grabbing using libdc1394
> and libraw1394 [no]
> @@ -1124,6 +1125,7 @@ EXTERNAL_LIBRARY_LIST="
> bzlib
> frei0r
> gnutls
> + libbs2b
> libcdio
> libdc1394
> libfaac
> @@ -2083,6 +2085,7 @@ unix_protocol_select="network"
> # filters
> blackframe_filter_deps="gpl"
> boxblur_filter_deps="gpl"
> +bs2b_filter_deps="libbs2b"
> cropdetect_filter_deps="gpl"
> delogo_filter_deps="gpl"
> drawtext_filter_deps="libfreetype"
> @@ -4024,6 +4027,7 @@ enabled avisynth && { { check_header
> "avisynth/avisynth_c.h" && check_l
> die "ERROR: LoadLibrary/dlopen not found, or
> avisynth header not found"; }
> enabled frei0r && { check_header frei0r.h || die "ERROR: frei0r.h
> header not found"; }
> enabled gnutls && require_pkg_config gnutls gnutls/gnutls.h
> gnutls_global_init
> +enabled libbs2b && require_pkg_config libbs2b bs2b.h bs2b_open
> enabled libfaac && require2 libfaac "stdint.h faac.h"
> faacEncGetVersion -lfaac
> enabled libfdk_aac && require libfdk_aac fdk-aac/aacenc_lib.h
> aacEncOpen -lfdk-aac
> enabled libfontconfig && require_pkg_config fontconfig
> "fontconfig/fontconfig.h" FcInit
> diff --git a/doc/filters.texi b/doc/filters.texi
> index d10a107..5f9d1f8 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -445,6 +445,36 @@ avconv -i INPUT -af atrim=end_sample=1000
>
> @end itemize
>
> +@section bs2b
> +Bauer stereo to binaural transformation, which improves headphone listening
> of
> +stereo audio records.
> +
> +It accepts the following parameters:
> +@table @option
> +
> +@item profile
> +Pre-defined crossfeed level.
> +@table @option
> +
> +@item default
> +Default level (fcut=700, feed=50).
> +
> +@item cmoy
> +Chu Moy circuit (fcut=700, feed=60).
> +
> +@item jmeier
> +Jan Meier circuit (fcut=650, feed=95).
> +
> +@end table
> +
> +@item fcut
> +Cut frequency (in Hz).
> +
> +@item feed
> +Feed level (in Hz).
> +
> +@end table
> +
> @section channelsplit
> Split each channel from an input audio stream into a separate output stream.
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 9c5c666..6537a8a 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -33,6 +33,7 @@ OBJS-$(CONFIG_ASHOWINFO_FILTER) +=
> af_ashowinfo.o
> OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
> OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
> OBJS-$(CONFIG_ATRIM_FILTER) += trim.o
> +OBJS-$(CONFIG_BS2B_FILTER) += af_bs2b.o
> OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
> OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
> OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
> diff --git a/libavfilter/af_bs2b.c b/libavfilter/af_bs2b.c
> new file mode 100644
> index 0000000..dc8b87f
> --- /dev/null
> +++ b/libavfilter/af_bs2b.c
> @@ -0,0 +1,213 @@
> +/*
> + * 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
> + * Bauer stereo-to-binaural filter
> + */
> +
> +#include <bs2b.h>
> +
> +#include "libavutil/channel_layout.h"
> +#include "libavutil/common.h"
> +#include "libavutil/float_dsp.h"
> +#include "libavutil/opt.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +typedef struct Bs2bContext {
> + const AVClass *class;
> + AVFloatDSPContext fdsp;
> +
> + int profile;
> + int fcut;
> + int feed;
> +
> + t_bs2bdp bs2bp;
> +
> + void (*filter)(t_bs2bdp bs2bdp, uint8_t *sample, int n);
> +} Bs2bContext;
> +
> +#define OFFSET(x) offsetof(Bs2bContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM
> +
> +static const AVOption options[] = {
> + { "profile", "Apply a pre-defined crossfeed level",
> + OFFSET(profile), AV_OPT_TYPE_INT, { .i64 = BS2B_DEFAULT_CLEVEL
> }, 0, INT_MAX, A, "profile" },
> + { "default", "default profile", 0, AV_OPT_TYPE_CONST, { .i64 =
> BS2B_DEFAULT_CLEVEL }, 0, 0, A, "profile" },
> + { "cmoy", "Chu Moy circuit", 0, AV_OPT_TYPE_CONST, { .i64 =
> BS2B_CMOY_CLEVEL }, 0, 0, A, "profile" },
> + { "jmeier", "Jan Meier circuit", 0, AV_OPT_TYPE_CONST, { .i64 =
> BS2B_JMEIER_CLEVEL }, 0, 0, A, "profile" },
> + { "fcut", "Set cut frequency (in Hz)",
> + OFFSET(fcut), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFCUT, A
> },
> + { "feed", "Set feed level (in Hz)",
> + OFFSET(feed), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, BS2B_MAXFEED, A
> },
> + { NULL },
> +};
> +
> +static const AVClass bs2b_class = {
> + .class_name = "bs2b filter",
> + .item_name = av_default_item_name,
> + .option = options,
> + .version = LIBAVUTIL_VERSION_INT,
> +};
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> + Bs2bContext *bs2b = ctx->priv;
> +
> + if (!(bs2b->bs2bp = bs2b_open()))
> + return -1;
AVERROR_UNKNOWN or maybe AVERROR(ENOMEM) ?
> + bs2b_set_level(bs2b->bs2bp, bs2b->profile);
> +
> + if (bs2b->fcut)
> + bs2b_set_level_fcut(bs2b->bs2bp, bs2b->fcut);
> +
> + if (bs2b->feed)
> + bs2b_set_level_feed(bs2b->bs2bp, bs2b->feed);
> +
> + return 0;
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats = NULL;
> + AVFilterChannelLayouts *layouts;
> +
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_U8,
> + AV_SAMPLE_FMT_S16,
> + AV_SAMPLE_FMT_S32,
> + AV_SAMPLE_FMT_FLT,
> + AV_SAMPLE_FMT_DBL,
> + AV_SAMPLE_FMT_NONE,
> + };
> +
> + layouts = ff_all_channel_layouts();
> + ff_add_channel_layout(&layouts, av_get_default_channel_layout(2));
> + 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 int filter_frame(AVFilterLink *inlink, AVFrame *buf)
> +{
> + int ret;
> + AVFrame *out_buf;
> +
> + Bs2bContext *bs2b = inlink->dst->priv;
> + AVFilterLink *outlink = inlink->dst->outputs[0];
> +
> + if (av_frame_is_writable(buf)) {
> + out_buf = buf;
> + } else {
> + out_buf = ff_get_audio_buffer(inlink, buf->nb_samples);
> + if (!out_buf)
> + return AVERROR(ENOMEM);
> + ret = av_frame_copy_props(out_buf, buf);
> + if (ret < 0) {
> + av_frame_free(&out_buf);
> + av_frame_free(&buf);
Not sure about this free, if you could fix it and the other small issues
below would be great.
> + return ret;
> + }
> + }
> +
> + bs2b->filter(bs2b->bs2bp, out_buf->extended_data[0],
> out_buf->nb_samples);
> +
> + return ff_filter_frame(outlink, out_buf);
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + Bs2bContext *bs2b = ctx->priv;
> + AVFilterLink *inlink = ctx->inputs[0];
> +
> + int srate = inlink->sample_rate;
> +
> + switch (av_get_packed_sample_fmt(inlink->format)) {
> + case AV_SAMPLE_FMT_U8:
> + bs2b->filter = bs2b_cross_feed_u8;
> + break;
> + case AV_SAMPLE_FMT_S16:
> + bs2b->filter = bs2b_cross_feed_s16;
> + break;
> + case AV_SAMPLE_FMT_S32:
> + bs2b->filter = bs2b_cross_feed_s32;
> + break;
> + case AV_SAMPLE_FMT_FLT:
> + bs2b->filter = bs2b_cross_feed_f;
> + break;
> + case AV_SAMPLE_FMT_DBL:
> + bs2b->filter = bs2b_cross_feed_d;
> + break;
> + default:
> + return -1;
AVERROR(ENOSYS);
> + }
> +
> + if ((srate < BS2B_MINSRATE) || (srate > BS2B_MAXSRATE))
> + return -1;
AVERROR(ENOSYS);
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