generic-library/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
96d616052b
ffmpeg/libavfilter/af_volume.c
Ganesh Ajjanagadde db1a642cd2 all: move ff_exp10, ff_exp10f, ff_fast_powf to lavu/ffmath.h 
The idea is to use ffmath.h for internal implementations of math functions.
Currently, it is used for variants of libm functions, but is by no means
limited to such things.

Note that this is not exported; use lavu/mathematics for such purposes.

Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanag@gmail.com>
2016-03-22 10:15:31 -07:00

494 lines
17 KiB
C
Raw Blame History

/*
* 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
static const AVOption volume_options[] = {
{ "volume", "set volume adjustment expression",
OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = 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" },
{ "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;
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
}
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts[vol->precision]);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
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;
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;
av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
}
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;
}
#define D2TS(d) (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
#define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
#define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
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);
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;
pos = av_frame_get_pkt_pos(buf);
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(inlink, nb_samples);
if (!out_buf)
return AVERROR(ENOMEM);
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,
},
{ NULL }
};
static const AVFilterPad avfilter_af_volume_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};
AVFilter ff_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,
.uninit = uninit,
.inputs = avfilter_af_volume_inputs,
.outputs = avfilter_af_volume_outputs,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
.process_command = process_command,
};
Reference in New Issue View Git Blame Copy Permalink