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FFmpeg/libavfilter/af_volume.c

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/*
* 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/ffmath.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/replaygain.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "af_volume.h"
static const char * const precision_str[] = {
"fixed", "float", "double"
};
static const char *const var_names[] = {
"n", ///< frame number (starting at zero)
"nb_channels", ///< number of channels
"nb_consumed_samples", ///< number of samples consumed by the filter
"nb_samples", ///< number of samples in the current frame
"pos", ///< position in the file of the frame
"pts", ///< frame presentation timestamp
"sample_rate", ///< sample rate
"startpts", ///< PTS at start of stream
"startt", ///< time at start of stream
"t", ///< time in the file of the frame
"tb", ///< timebase
"volume", ///< last set value
NULL
};
#define OFFSET(x) offsetof(VolumeContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
#define T AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption volume_options[] = {
{ "volume", "set volume adjustment expression",
OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F|T },
{ "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" },
{ "eval", "specify when to evaluate expressions", OFFSET(eval_mode), AV_OPT_TYPE_INT, {.i64 = EVAL_MODE_ONCE}, 0, EVAL_MODE_NB-1, .flags = A|F, "eval" },
{ "once", "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE}, .flags = A|F, .unit = "eval" },
{ "frame", "eval volume expression per-frame", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },
{ "replaygain", "Apply replaygain side data when present",
OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A|F, "replaygain" },
{ "drop", "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP }, 0, 0, A|F, "replaygain" },
{ "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A|F, "replaygain" },
{ "track", "track gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK }, 0, 0, A|F, "replaygain" },
{ "album", "album gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM }, 0, 0, A|F, "replaygain" },
{ "replaygain_preamp", "Apply replaygain pre-amplification",
OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A|F },
{ "replaygain_noclip", "Apply replaygain clipping prevention",
OFFSET(replaygain_noclip), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, A|F },
{ NULL }
};
AVFILTER_DEFINE_CLASS(volume);
static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
{
int ret;
AVExpr *old = NULL;
if (*pexpr)
old = *pexpr;
ret = av_expr_parse(pexpr, expr, var_names,
NULL, NULL, NULL, NULL, 0, log_ctx);
if (ret < 0) {
av_log(log_ctx, AV_LOG_ERROR,
"Error when evaluating the volume expression '%s'\n", expr);
*pexpr = old;
return ret;
}
av_expr_free(old);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
VolumeContext *vol = ctx->priv;
vol->fdsp = avpriv_float_dsp_alloc(0);
if (!vol->fdsp)
return AVERROR(ENOMEM);
return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
}
static av_cold void uninit(AVFilterContext *ctx)
{
VolumeContext *vol = ctx->priv;
av_expr_free(vol->volume_pexpr);
av_opt_free(vol);
av_freep(&vol->fdsp);
}
static int query_formats(AVFilterContext *ctx)
{
VolumeContext *vol = ctx->priv;
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
}
};
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts[vol->precision]);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
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:
vol->samples_align = 4;
break;
case AV_SAMPLE_FMT_DBL:
vol->samples_align = 8;
break;
}
if (ARCH_X86)
ff_volume_init_x86(vol);
}
static int set_volume(AVFilterContext *ctx)
{
VolumeContext *vol = ctx->priv;
vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
if (isnan(vol->volume)) {
if (vol->eval_mode == EVAL_MODE_ONCE) {
av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
return AVERROR(EINVAL);
} else {
av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
vol->volume = 0;
}
}
vol->var_values[VAR_VOLUME] = vol->volume;
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av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
precision_str[vol->precision]);
if (vol->precision == PRECISION_FIXED) {
vol->volume_i = (int)(vol->volume * 256 + 0.5);
vol->volume = vol->volume_i / 256.0;
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av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
}
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av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
vol->volume, 20.0*log10(vol->volume));
volume_init(vol);
return 0;
}
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 = inlink->channels;
vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
vol->var_values[VAR_N] =
vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
vol->var_values[VAR_NB_SAMPLES] =
vol->var_values[VAR_POS] =
vol->var_values[VAR_PTS] =
vol->var_values[VAR_STARTPTS] =
vol->var_values[VAR_STARTT] =
vol->var_values[VAR_T] =
vol->var_values[VAR_VOLUME] = NAN;
vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
vol->var_values[VAR_TB] = av_q2d(inlink->time_base);
vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
vol->var_values[VAR_TB],
vol->var_values[VAR_SAMPLE_RATE],
vol->var_values[VAR_NB_CHANNELS]);
return set_volume(ctx);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
VolumeContext *vol = ctx->priv;
int ret = AVERROR(ENOSYS);
if (!strcmp(cmd, "volume")) {
if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
return ret;
if (vol->eval_mode == EVAL_MODE_ONCE)
set_volume(ctx);
}
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
int nb_samples = buf->nb_samples;
AVFrame *out_buf;
int64_t pos;
AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
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int ret;
if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
if (vol->replaygain != REPLAYGAIN_DROP) {
AVReplayGain *replaygain = (AVReplayGain*)sd->data;
int32_t gain = 100000;
uint32_t peak = 100000;
float g, p;
if (vol->replaygain == REPLAYGAIN_TRACK &&
replaygain->track_gain != INT32_MIN) {
gain = replaygain->track_gain;
if (replaygain->track_peak != 0)
peak = replaygain->track_peak;
} else if (replaygain->album_gain != INT32_MIN) {
gain = replaygain->album_gain;
if (replaygain->album_peak != 0)
peak = replaygain->album_peak;
} else {
av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
"values are unknown.\n");
}
g = gain / 100000.0f;
p = peak / 100000.0f;
av_log(inlink->dst, AV_LOG_VERBOSE,
"Using gain %f dB from replaygain side data.\n", g);
vol->volume = ff_exp10((g + vol->replaygain_preamp) / 20);
if (vol->replaygain_noclip)
vol->volume = FFMIN(vol->volume, 1.0 / p);
vol->volume_i = (int)(vol->volume * 256 + 0.5);
volume_init(vol);
}
av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
}
if (isnan(vol->var_values[VAR_STARTPTS])) {
vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
vol->var_values[VAR_STARTT ] = TS2T(buf->pts, inlink->time_base);
}
vol->var_values[VAR_PTS] = TS2D(buf->pts);
vol->var_values[VAR_T ] = TS2T(buf->pts, inlink->time_base);
vol->var_values[VAR_N ] = inlink->frame_count_out;
pos = buf->pkt_pos;
vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
if (vol->eval_mode == EVAL_MODE_FRAME)
set_volume(ctx);
if (vol->volume == 1.0 || vol->volume_i == 256) {
out_buf = buf;
goto end;
}
/* do volume scaling in-place if input buffer is writable */
if (av_frame_is_writable(buf)
&& (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
out_buf = buf;
} else {
out_buf = ff_get_audio_buffer(outlink, nb_samples);
if (!out_buf) {
av_frame_free(&buf);
return AVERROR(ENOMEM);
}
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ret = av_frame_copy_props(out_buf, buf);
if (ret < 0) {
av_frame_free(&out_buf);
av_frame_free(&buf);
return ret;
}
}
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);
}
}
}
emms_c();
if (buf != out_buf)
av_frame_free(&buf);
end:
vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;
return ff_filter_frame(outlink, out_buf);
}
static const AVFilterPad avfilter_af_volume_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad avfilter_af_volume_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_volume = {
.name = "volume",
.description = NULL_IF_CONFIG_SMALL("Change input volume."),
.priv_size = sizeof(VolumeContext),
.priv_class = &volume_class,
.init = init,
.uninit = uninit,
2021-08-12 13:05:31 +02:00
FILTER_INPUTS(avfilter_af_volume_inputs),
FILTER_OUTPUTS(avfilter_af_volume_outputs),
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 12:07:35 +02:00
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
.process_command = process_command,
};