From edf217ebb7d518be3030184d03b5534033e82d0f Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Wed, 28 Jan 2015 15:46:58 +0000 Subject: avfilter: add dcshift filter Signed-off-by: Paul B Mahol --- Changelog | 1 + doc/filters.texi | 19 ++++++ libavfilter/Makefile | 1 + libavfilter/af_dcshift.c | 164 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + libavfilter/version.h | 4 +- 6 files changed, 188 insertions(+), 2 deletions(-) create mode 100644 libavfilter/af_dcshift.c diff --git a/Changelog b/Changelog index 49b4793f01..c663d5eb84 100644 --- a/Changelog +++ b/Changelog @@ -22,6 +22,7 @@ version : - removed libmpcodecs - Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range - ported softpulldown filter from libmpcodecs as repeatfields filter +- dcshift filter version 2.5: diff --git a/doc/filters.texi b/doc/filters.texi index 2f29c46f9c..8069554eb4 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -917,6 +917,7 @@ audio, the data is treated as if all the planes were concatenated. A list of Adler-32 checksums for each data plane. @end table +@anchor{astats} @section astats Display time domain statistical information about the audio channels. @@ -1394,6 +1395,24 @@ compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1 @end example @end itemize +@section dcshift +Apply a DC shift to the audio. + +This can be useful to remove a DC offset (caused perhaps by a hardware problem +in the recording chain) from the audio. The effect of a DC offset is reduced +headroom and hence volume. The @ref{astats} filter can be used to determine if +a signal has a DC offset. + +@table @option +@item shift +Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift +the audio. + +@item limitergain +Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is +used to prevent clipping. +@end table + @section earwax Make audio easier to listen to on headphones. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 21a3fbe7ce..7d1ea9199b 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -65,6 +65,7 @@ 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 +OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o diff --git a/libavfilter/af_dcshift.c b/libavfilter/af_dcshift.c new file mode 100644 index 0000000000..c1abb3c365 --- /dev/null +++ b/libavfilter/af_dcshift.c @@ -0,0 +1,164 @@ +/* + * Copyright (c) 2000 Chris Ausbrooks + * Copyright (c) 2000 Fabien COELHO + * + * 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 + */ + +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" +#include "avfilter.h" +#include "audio.h" +#include "internal.h" + +typedef struct DCShiftContext { + const AVClass *class; + double dcshift; + double limiterthreshhold; + double limitergain; +} DCShiftContext; + +#define OFFSET(x) offsetof(DCShiftContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption dcshift_options[] = { + { "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A }, + { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(dcshift); + +static av_cold int init(AVFilterContext *ctx) +{ + DCShiftContext *s = ctx->priv; + + s->limiterthreshhold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain)); + + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterChannelLayouts *layouts; + AVFilterFormats *formats; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_S32P, 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 int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples); + DCShiftContext *s = ctx->priv; + int i, j; + double dcshift = s->dcshift; + + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + + if (s->limitergain > 0) { + for (i = 0; i < inlink->channels; i++) { + const int32_t *src = (int32_t *)in->extended_data[i]; + int32_t *dst = (int32_t *)out->extended_data[i]; + + for (j = 0; j < in->nb_samples; j++) { + double d; + + d = src[j]; + + if (d > s->limiterthreshhold && dcshift > 0) { + d = (d - s->limiterthreshhold) * s->limitergain / + (INT32_MAX - s->limiterthreshhold) + + s->limiterthreshhold + dcshift; + } else if (d < -s->limiterthreshhold && dcshift < 0) { + d = (d + s->limiterthreshhold) * s->limitergain / + (INT32_MAX - s->limiterthreshhold) - + s->limiterthreshhold + dcshift; + } else { + d = dcshift * INT32_MAX + d; + } + + dst[j] = av_clipl_int32(d); + } + } + } else { + for (i = 0; i < inlink->channels; i++) { + const int32_t *src = (int32_t *)in->extended_data[i]; + int32_t *dst = (int32_t *)out->extended_data[i]; + + for (j = 0; j < in->nb_samples; j++) { + double d = dcshift * (INT32_MAX + 1.) + src[j]; + + dst[j] = av_clipl_int32(d); + } + } + } + + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} +static const AVFilterPad dcshift_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad dcshift_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_dcshift = { + .name = "dcshift", + .description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."), + .query_formats = query_formats, + .priv_size = sizeof(DCShiftContext), + .priv_class = &dcshift_class, + .init = init, + .inputs = dcshift_inputs, + .outputs = dcshift_outputs, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 9c6f2aeb0c..62d3eb33ce 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -81,6 +81,7 @@ void avfilter_register_all(void) REGISTER_FILTER(CHANNELMAP, channelmap, af); REGISTER_FILTER(CHANNELSPLIT, channelsplit, af); REGISTER_FILTER(COMPAND, compand, af); + REGISTER_FILTER(DCSHIFT, dcshift, af); REGISTER_FILTER(EARWAX, earwax, af); REGISTER_FILTER(EBUR128, ebur128, af); REGISTER_FILTER(EQUALIZER, equalizer, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 49ea3a90da..4e506888e0 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,8 +30,8 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 5 -#define LIBAVFILTER_VERSION_MINOR 9 -#define LIBAVFILTER_VERSION_MICRO 104 +#define LIBAVFILTER_VERSION_MINOR 10 +#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \ -- cgit v1.2.3