| 0,0 → 1,300 |
|---|
| /* |
| * Copyright (c) 2011 Stefano Sabatini |
| * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
| * |
| * 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 |
| */ |
| |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/common.h" |
| #include "libavutil/eval.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/opt.h" |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "formats.h" |
| #include "internal.h" |
| #include "af_volume.h" |
| |
| static const char *precision_str[] = { |
| "fixed", "float", "double" |
| }; |
| |
| #define OFFSET(x) offsetof(VolumeContext, x) |
| #define A AV_OPT_FLAG_AUDIO_PARAM |
| #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 } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(volume); |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| VolumeContext *vol = ctx->priv; |
| |
| if (vol->precision == PRECISION_FIXED) { |
| vol->volume_i = (int)(vol->volume * 256 + 0.5); |
| vol->volume = vol->volume_i / 256.0; |
| av_log(ctx, AV_LOG_VERBOSE, "volume:(%d/256)(%f)(%1.2fdB) precision:fixed\n", |
| vol->volume_i, vol->volume, 20.0*log(vol->volume)/M_LN10); |
| } else { |
| av_log(ctx, AV_LOG_VERBOSE, "volume:(%f)(%1.2fdB) precision:%s\n", |
| vol->volume, 20.0*log(vol->volume)/M_LN10, |
| precision_str[vol->precision]); |
| } |
| |
| return 0; |
| } |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| VolumeContext *vol = ctx->priv; |
| AVFilterFormats *formats = NULL; |
| AVFilterChannelLayouts *layouts; |
| static const enum AVSampleFormat sample_fmts[][7] = { |
| [PRECISION_FIXED] = { |
| AV_SAMPLE_FMT_U8, |
| AV_SAMPLE_FMT_U8P, |
| AV_SAMPLE_FMT_S16, |
| AV_SAMPLE_FMT_S16P, |
| AV_SAMPLE_FMT_S32, |
| AV_SAMPLE_FMT_S32P, |
| AV_SAMPLE_FMT_NONE |
| }, |
| [PRECISION_FLOAT] = { |
| AV_SAMPLE_FMT_FLT, |
| AV_SAMPLE_FMT_FLTP, |
| AV_SAMPLE_FMT_NONE |
| }, |
| [PRECISION_DOUBLE] = { |
| AV_SAMPLE_FMT_DBL, |
| AV_SAMPLE_FMT_DBLP, |
| 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[vol->precision]); |
| 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 inline void scale_samples_u8(uint8_t *dst, const uint8_t *src, |
| int nb_samples, int volume) |
| { |
| int i; |
| for (i = 0; i < nb_samples; i++) |
| dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128); |
| } |
| |
| static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src, |
| int nb_samples, int volume) |
| { |
| int i; |
| for (i = 0; i < nb_samples; i++) |
| dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128); |
| } |
| |
| static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src, |
| int nb_samples, int volume) |
| { |
| int i; |
| int16_t *smp_dst = (int16_t *)dst; |
| const int16_t *smp_src = (const int16_t *)src; |
| for (i = 0; i < nb_samples; i++) |
| smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8); |
| } |
| |
| static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src, |
| int nb_samples, int volume) |
| { |
| int i; |
| int16_t *smp_dst = (int16_t *)dst; |
| const int16_t *smp_src = (const int16_t *)src; |
| for (i = 0; i < nb_samples; i++) |
| smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8); |
| } |
| |
| static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src, |
| int nb_samples, int volume) |
| { |
| int i; |
| int32_t *smp_dst = (int32_t *)dst; |
| const int32_t *smp_src = (const int32_t *)src; |
| for (i = 0; i < nb_samples; i++) |
| 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; |
| |
| switch (av_get_packed_sample_fmt(vol->sample_fmt)) { |
| case AV_SAMPLE_FMT_U8: |
| if (vol->volume_i < 0x1000000) |
| vol->scale_samples = scale_samples_u8_small; |
| else |
| vol->scale_samples = scale_samples_u8; |
| break; |
| case AV_SAMPLE_FMT_S16: |
| if (vol->volume_i < 0x10000) |
| vol->scale_samples = scale_samples_s16_small; |
| else |
| vol->scale_samples = scale_samples_s16; |
| break; |
| case AV_SAMPLE_FMT_S32: |
| vol->scale_samples = scale_samples_s32; |
| break; |
| case AV_SAMPLE_FMT_FLT: |
| avpriv_float_dsp_init(&vol->fdsp, 0); |
| vol->samples_align = 4; |
| break; |
| case AV_SAMPLE_FMT_DBL: |
| avpriv_float_dsp_init(&vol->fdsp, 0); |
| vol->samples_align = 8; |
| break; |
| } |
| |
| if (ARCH_X86) |
| ff_volume_init_x86(vol); |
| } |
| |
| static int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| VolumeContext *vol = ctx->priv; |
| AVFilterLink *inlink = ctx->inputs[0]; |
| |
| vol->sample_fmt = inlink->format; |
| vol->channels = av_get_channel_layout_nb_channels(inlink->channel_layout); |
| vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1; |
| |
| volume_init(vol); |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *buf) |
| { |
| VolumeContext *vol = inlink->dst->priv; |
| AVFilterLink *outlink = inlink->dst->outputs[0]; |
| int nb_samples = buf->nb_samples; |
| AVFrame *out_buf; |
| |
| if (vol->volume == 1.0 || vol->volume_i == 256) |
| return ff_filter_frame(outlink, buf); |
| |
| /* do volume scaling in-place if input buffer is writable */ |
| if (av_frame_is_writable(buf)) { |
| out_buf = buf; |
| } else { |
| out_buf = ff_get_audio_buffer(inlink, nb_samples); |
| if (!out_buf) |
| return AVERROR(ENOMEM); |
| av_frame_copy_props(out_buf, buf); |
| } |
| |
| if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) { |
| int p, plane_samples; |
| |
| if (av_sample_fmt_is_planar(buf->format)) |
| plane_samples = FFALIGN(nb_samples, vol->samples_align); |
| else |
| plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align); |
| |
| if (vol->precision == PRECISION_FIXED) { |
| for (p = 0; p < vol->planes; p++) { |
| vol->scale_samples(out_buf->extended_data[p], |
| buf->extended_data[p], plane_samples, |
| vol->volume_i); |
| } |
| } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) { |
| for (p = 0; p < vol->planes; p++) { |
| vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p], |
| (const float *)buf->extended_data[p], |
| vol->volume, plane_samples); |
| } |
| } else { |
| for (p = 0; p < vol->planes; p++) { |
| vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p], |
| (const double *)buf->extended_data[p], |
| vol->volume, plane_samples); |
| } |
| } |
| } |
| |
| if (buf != out_buf) |
| av_frame_free(&buf); |
| |
| return ff_filter_frame(outlink, out_buf); |
| } |
| |
| static const AVFilterPad avfilter_af_volume_inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .filter_frame = filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad avfilter_af_volume_outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter avfilter_af_volume = { |
| .name = "volume", |
| .description = NULL_IF_CONFIG_SMALL("Change input volume."), |
| .query_formats = query_formats, |
| .priv_size = sizeof(VolumeContext), |
| .priv_class = &volume_class, |
| .init = init, |
| .inputs = avfilter_af_volume_inputs, |
| .outputs = avfilter_af_volume_outputs, |
| .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, |
| }; |