[FFmpeg-cvslog] lavfi: add Bauer stereo-to-binaural audio filter

Alessandro Ghedini git at videolan.org
Thu May 1 14:28:25 CEST 2014


ffmpeg | branch: master | Alessandro Ghedini <alessandro at ghedini.me> | Tue Apr 29 18:53:16 2014 +0200| [1c0210c7981b6a61043d9171f506b435ff5a1f5e] | committer: Anton Khirnov

lavfi: add Bauer stereo-to-binaural audio filter

Signed-off-by: Anton Khirnov <anton at khirnov.net>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1c0210c7981b6a61043d9171f506b435ff5a1f5e
---

 Changelog                |    1 +
 configure                |    4 +
 doc/filters.texi         |   30 +++++++
 libavfilter/Makefile     |    1 +
 libavfilter/af_bs2b.c    |  222 ++++++++++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |    1 +
 libavfilter/version.h    |    2 +-
 7 files changed, 260 insertions(+), 1 deletion(-)

diff --git a/Changelog b/Changelog
index eddcf1d..2dac228 100644
--- a/Changelog
+++ b/Changelog
@@ -24,6 +24,7 @@ version <next>:
 - Silicon Graphics Movie demuxer
 - On2 AVC (Audio for Video) decoder
 - support for decoding through DXVA2 in avconv
+- libbs2b-based stereo-to-binaural audio filter
 
 
 version 10:
diff --git a/configure b/configure
index 30f90eb..47fd690 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
@@ -2085,6 +2087,7 @@ 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"
@@ -4027,6 +4030,7 @@ 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
 
+ at section bs2b
+Bauer stereo to binaural transformation, which improves headphone listening of
+stereo audio records.
+
+It accepts the following parameters:
+ at table @option
+
+ at item profile
+Pre-defined crossfeed level.
+ at table @option
+
+ at item default
+Default level (fcut=700, feed=50).
+
+ at item cmoy
+Chu Moy circuit (fcut=700, feed=60).
+
+ at item jmeier
+Jan Meier circuit (fcut=650, feed=95).
+
+ at end table
+
+ at item fcut
+Cut frequency (in Hz).
+
+ at item feed
+Feed level (in Hz).
+
+ at 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..25e7867
--- /dev/null
+++ b/libavfilter/af_bs2b.c
@@ -0,0 +1,222 @@
+/*
+ * 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/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+typedef struct Bs2bContext {
+    const AVClass *class;
+
+    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 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 av_cold void uninit(AVFilterContext *ctx)
+{
+    Bs2bContext *bs2b = ctx->priv;
+
+    if (bs2b->bs2bp)
+        bs2b_close(bs2b->bs2bp);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats = NULL;
+    AVFilterChannelLayouts *layouts = NULL;
+
+    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,
+    };
+
+    if (ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO) != 0)
+        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 *frame)
+{
+    int ret;
+    AVFrame *out_frame;
+
+    Bs2bContext     *bs2b = inlink->dst->priv;
+    AVFilterLink *outlink = inlink->dst->outputs[0];
+
+    if (av_frame_is_writable(frame)) {
+        out_frame = frame;
+    } else {
+        out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+        if (!out_frame)
+            return AVERROR(ENOMEM);
+        av_frame_copy(out_frame, frame);
+        ret = av_frame_copy_props(out_frame, frame);
+        if (ret < 0) {
+            av_frame_free(&out_frame);
+            av_frame_free(&frame);
+            return ret;
+        }
+    }
+
+    bs2b->filter(bs2b->bs2bp, out_frame->extended_data[0], out_frame->nb_samples);
+
+    if (frame != out_frame)
+        av_frame_free(&frame);
+
+    return ff_filter_frame(outlink, out_frame);
+}
+
+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 (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 AVERROR_BUG;
+    }
+
+    if ((srate < BS2B_MINSRATE) || (srate > BS2B_MAXSRATE))
+        return AVERROR(ENOSYS);
+
+    bs2b_set_srate(bs2b->bs2bp, srate);
+
+    return 0;
+}
+
+static const AVFilterPad bs2b_inputs[] = {
+    {
+        .name           = "default",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .filter_frame   = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad bs2b_outputs[] = {
+    {
+        .name           = "default",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .config_props   = config_output,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_bs2b = {
+    .name           = "bs2b",
+    .description    = NULL_IF_CONFIG_SMALL("Bauer stereo-to-binaural filter."),
+    .query_formats  = query_formats,
+    .priv_size      = sizeof(Bs2bContext),
+    .priv_class     = &bs2b_class,
+    .init           = init,
+    .uninit         = uninit,
+    .inputs         = bs2b_inputs,
+    .outputs        = bs2b_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d7bb47a..67a298d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -53,6 +53,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ASPLIT,         asplit,         af);
     REGISTER_FILTER(ASYNCTS,        asyncts,        af);
     REGISTER_FILTER(ATRIM,          atrim,          af);
+    REGISTER_FILTER(BS2B,           bs2b,           af);
     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
     REGISTER_FILTER(COMPAND,        compand,        af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 5093e50..de64b8b 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  4
-#define LIBAVFILTER_VERSION_MINOR  4
+#define LIBAVFILTER_VERSION_MINOR  5
 #define LIBAVFILTER_VERSION_MICRO  0
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \



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