diff options
Diffstat (limited to 'libavfilter/af_volume.c')
| -rw-r--r-- | libavfilter/af_volume.c | 50 |
1 files changed, 21 insertions, 29 deletions
diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c index 85d7067132..cd248772e5 100644 --- a/libavfilter/af_volume.c +++ b/libavfilter/af_volume.c @@ -2,20 +2,20 @@ * Copyright (c) 2011 Stefano Sabatini * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -41,24 +41,20 @@ static const char *precision_str[] = { #define OFFSET(x) offsetof(VolumeContext, x) #define A AV_OPT_FLAG_AUDIO_PARAM - -static const AVOption options[] = { - { "volume", "Volume adjustment.", - OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A }, - { "precision", "Mathematical precision.", - OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A, "precision" }, - { "fixed", "8-bit fixed-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A, "precision" }, - { "float", "32-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A, "precision" }, - { "double", "64-bit floating-point.", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A, "precision" }, - { NULL }, +#define F AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption volume_options[] = { + { "volume", "set volume adjustment", + OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 1.0 }, 0, 0x7fffff, A|F }, + { "precision", "select mathematical precision", + OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" }, + { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" }, + { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" }, + { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" }, + { NULL } }; -static const AVClass volume_class = { - .class_name = "volume filter", - .item_name = av_default_item_name, - .option = options, - .version = LIBAVUTIL_VERSION_INT, -}; +AVFILTER_DEFINE_CLASS(volume); static av_cold int init(AVFilterContext *ctx) { @@ -84,8 +80,7 @@ static int query_formats(AVFilterContext *ctx) AVFilterFormats *formats = NULL; AVFilterChannelLayouts *layouts; static const enum AVSampleFormat sample_fmts[][7] = { - /* PRECISION_FIXED */ - { + [PRECISION_FIXED] = { AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16, @@ -94,14 +89,12 @@ static int query_formats(AVFilterContext *ctx) AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE }, - /* PRECISION_FLOAT */ - { + [PRECISION_FLOAT] = { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, - /* PRECISION_DOUBLE */ - { + [PRECISION_DOUBLE] = { AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE @@ -172,8 +165,6 @@ static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src, smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8)); } - - static av_cold void volume_init(VolumeContext *vol) { vol->samples_align = 1; @@ -240,7 +231,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf) out_buf = ff_get_audio_buffer(inlink, nb_samples); if (!out_buf) return AVERROR(ENOMEM); - out_buf->pts = buf->pts; + av_frame_copy_props(out_buf, buf); } if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) { @@ -305,4 +296,5 @@ AVFilter ff_af_volume = { .init = init, .inputs = avfilter_af_volume_inputs, .outputs = avfilter_af_volume_outputs, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, }; |