diff options
Diffstat (limited to 'libavfilter/af_volume.c')
| -rw-r--r-- | libavfilter/af_volume.c | 191 |
1 files changed, 191 insertions, 0 deletions
diff --git a/libavfilter/af_volume.c b/libavfilter/af_volume.c new file mode 100644 index 0000000..1aca7d9 --- /dev/null +++ b/libavfilter/af_volume.c @@ -0,0 +1,191 @@ +/* + * Copyright (c) 2011 Stefano Sabatini + * + * 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 + */ + +/** + * @file + * audio volume filter + * based on ffmpeg.c code + */ + +#include "libavutil/audioconvert.h" +#include "libavutil/eval.h" +#include "audio.h" +#include "avfilter.h" +#include "formats.h" + +typedef struct { + double volume; + int volume_i; +} VolumeContext; + +static av_cold int init(AVFilterContext *ctx, const char *args) +{ + VolumeContext *vol = ctx->priv; + char *tail; + int ret = 0; + + vol->volume = 1.0; + + if (args) { + /* parse the number as a decimal number */ + double d = strtod(args, &tail); + + if (*tail) { + if (!strcmp(tail, "dB")) { + /* consider the argument an adjustement in decibels */ + d = pow(10, d/20); + } else { + /* parse the argument as an expression */ + ret = av_expr_parse_and_eval(&d, args, NULL, NULL, + NULL, NULL, NULL, NULL, + NULL, 0, ctx); + } + } + + if (ret < 0) { + av_log(ctx, AV_LOG_ERROR, + "Invalid volume argument '%s'\n", args); + return AVERROR(EINVAL); + } + + if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */ + av_log(ctx, AV_LOG_ERROR, + "Negative or too big volume value %f\n", d); + return AVERROR(EINVAL); + } + + vol->volume = d; + } + + vol->volume_i = (int)(vol->volume * 256 + 0.5); + av_log(ctx, AV_LOG_VERBOSE, "volume=%f\n", vol->volume); + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts; + 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(); + 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_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +{ + VolumeContext *vol = inlink->dst->priv; + AVFilterLink *outlink = inlink->dst->outputs[0]; + const int nb_samples = insamples->audio->nb_samples * + av_get_channel_layout_nb_channels(insamples->audio->channel_layout); + const double volume = vol->volume; + const int volume_i = vol->volume_i; + int i; + + if (volume_i != 256) { + switch (insamples->format) { + case AV_SAMPLE_FMT_U8: + { + uint8_t *p = (void *)insamples->data[0]; + for (i = 0; i < nb_samples; i++) { + int v = (((*p - 128) * volume_i + 128) >> 8) + 128; + *p++ = av_clip_uint8(v); + } + break; + } + case AV_SAMPLE_FMT_S16: + { + int16_t *p = (void *)insamples->data[0]; + for (i = 0; i < nb_samples; i++) { + int v = ((int64_t)*p * volume_i + 128) >> 8; + *p++ = av_clip_int16(v); + } + break; + } + case AV_SAMPLE_FMT_S32: + { + int32_t *p = (void *)insamples->data[0]; + for (i = 0; i < nb_samples; i++) { + int64_t v = (((int64_t)*p * volume_i + 128) >> 8); + *p++ = av_clipl_int32(v); + } + break; + } + case AV_SAMPLE_FMT_FLT: + { + float *p = (void *)insamples->data[0]; + float scale = (float)volume; + for (i = 0; i < nb_samples; i++) { + *p++ *= scale; + } + break; + } + case AV_SAMPLE_FMT_DBL: + { + double *p = (void *)insamples->data[0]; + for (i = 0; i < nb_samples; i++) { + *p *= volume; + p++; + } + break; + } + } + } + return ff_filter_samples(outlink, insamples); +} + +AVFilter avfilter_af_volume = { + .name = "volume", + .description = NULL_IF_CONFIG_SMALL("Change input volume."), + .query_formats = query_formats, + .priv_size = sizeof(VolumeContext), + .init = init, + + .inputs = (const AVFilterPad[]) {{ .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_samples = filter_samples, + .min_perms = AV_PERM_READ|AV_PERM_WRITE}, + { .name = NULL}}, + + .outputs = (const AVFilterPad[]) {{ .name = "default", + .type = AVMEDIA_TYPE_AUDIO, }, + { .name = NULL}}, +}; |
