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avfilter/af_afir: add irnorm and irlink options

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Deprecate gtype option.
release/7.0
Paul B Mahol 1 year ago
parent 07c303b708
commit 5452cbdc15
4 changed files with 91 additions and 119 deletions
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  1. 37
      doc/filters.texi
  2. 57
      libavfilter/af_afir.c
  3. 3
      libavfilter/af_afir.h
  4. 113
      libavfilter/afir_template.c

37
doc/filters.texi
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@ -1794,33 +1794,24 @@ Set wet gain. This sets final output gain.
Set Impulse Response filter length. Default is 1, which means whole IR is processed.
@item gtype
Enable applying gain measured from power of IR.
This option is deprecated, and does nothing.
Set which approach to use for auto gain measurement.
@item irnorm
Set norm to be applied to IR coefficients before filtering.
Allowed range is from @var{-1} to @var{2}.
IR coefficients are normalized with calculated vector norm set by this option.
For negative values, no norm is calculated, and IR coefficients are not modified at all.
Default is @var{1}.
@table @option
@item none
Do not apply any gain.
@item peak
select peak gain, very conservative approach. This is default value.
@item dc
select DC gain, limited application.
@item gn
select gain to noise approach, this is most popular one.
@item ac
select AC gain.
@item rms
select RMS gain.
@end table
@item irlink
For multichannel IR if this option is set to @var{true}. All IR channels will be
normalized with maximal measured gain of all IR channels coefficients as set by @code{irnorm} option.
When disabled, all IR coefficients in each IR channel will be normalized independently.
Default is @var{true}.
@item irgain
Set gain to be applied to IR coefficients before filtering.
Allowed range is 0 to 1. This gain is applied after any gain applied with @var{gtype} option.
Allowed range is 0 to 1. This gain is applied after any gain applied with @var{irnorm} option.
@item irfmt
Set format of IR stream. Can be @code{mono} or @code{input}.
@ -1899,7 +1890,7 @@ ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
Apply true stereo processing given input stereo stream, and two stereo impulse responses for left and right channel,
the impulse response files are files with names l_ir.wav and r_ir.wav:
@example
"pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:gtype=gn:irgain=-5dB,pan=stereo|FL<c0+c2|FR<c1+c3"
"pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irgain=-5dB,pan=stereo|FL<c0+c2|FR<c1+c3"
@end example
@end itemize

57
libavfilter/af_afir.c
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@ -393,6 +393,22 @@ skip:
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
for (int ch = 0; ch < s->nb_channels; ch++) {
const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc);
}
if (s->ir_link) {
float gain = +INFINITY;
for (int ch = 0; ch < s->nb_channels; ch++)
gain = fminf(gain, s->ch_gain[ch]);
for (int ch = 0; ch < s->nb_channels; ch++)
s->ch_gain[ch] = gain;
}
for (int ch = 0; ch < s->nb_channels; ch++) {
const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch];
float *time = (float *)s->norm_ir[selir]->extended_data[ch];
@ -401,7 +417,7 @@ skip:
for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
time[i] = 0;
get_power_float(ctx, s, nb_taps, ch, time);
ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
for (int n = 0; n < s->nb_segments[selir]; n++) {
AudioFIRSegment *seg = &s->seg[selir][n];
@ -417,6 +433,22 @@ skip:
}
break;
case AV_SAMPLE_FMT_DBLP:
for (int ch = 0; ch < s->nb_channels; ch++) {
const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc);
}
if (s->ir_link) {
double gain = +INFINITY;
for (int ch = 0; ch < s->nb_channels; ch++)
gain = fmin(gain, s->ch_gain[ch]);
for (int ch = 0; ch < s->nb_channels; ch++)
s->ch_gain[ch] = gain;
}
for (int ch = 0; ch < s->nb_channels; ch++) {
const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch];
double *time = (double *)s->norm_ir[selir]->extended_data[ch];
@ -425,7 +457,8 @@ skip:
for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++)
time[i] = 0;
get_power_double(ctx, s, nb_taps, ch, time);
ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]);
for (int n = 0; n < s->nb_segments[selir]; n++) {
AudioFIRSegment *seg = &s->seg[selir][n];
@ -627,8 +660,9 @@ FF_ENABLE_DEPRECATION_WARNINGS
s->format = outlink->format;
s->nb_channels = outlink->ch_layout.nb_channels;
s->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain));
s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading));
if (!s->loading)
if (!s->loading || !s->ch_gain)
return AVERROR(ENOMEM);
s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
@ -674,6 +708,7 @@ static av_cold void uninit(AVFilterContext *ctx)
AudioFIRContext *s = ctx->priv;
av_freep(&s->fdsp);
av_freep(&s->ch_gain);
av_freep(&s->loading);
for (int i = 0; i < s->nb_irs; i++) {
@ -812,13 +847,15 @@ static const AVOption afir_options[] = {
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR },
{ "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF, "gtype" },
{ "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" },
{ "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" },
{ "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" },
{ "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" },
{ "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "gtype" },
{ "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "gtype" },
{ "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, "gtype" },
{ "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF },
{ "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
{ "irgain", "set IR gain", OFFSET(ir_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ "irfmt", "set IR format", OFFSET(ir_format), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, AF, "irfmt" },
{ "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "irfmt" },

3
libavfilter/af_afir.h
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@ -63,6 +63,8 @@ typedef struct AudioFIRContext {
float dry_gain;
float length;
int gtype;
float ir_norm;
float ir_link;
float ir_gain;
int ir_format;
int ir_load;
@ -87,6 +89,7 @@ typedef struct AudioFIRContext {
int nb_channels;
int one2many;
int *loading;
double *ch_gain;
AudioFIRSegment seg[MAX_IR_STREAMS][1024];

113
libavfilter/afir_template.c
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@ -29,6 +29,8 @@
#undef HYPOT
#undef SAMPLE_FORMAT
#undef TX_TYPE
#undef FABS
#undef POW
#if DEPTH == 32
#define SAMPLE_FORMAT float
#define SQRT sqrtf
@ -36,6 +38,8 @@
#define ctype AVComplexFloat
#define ftype float
#define TX_TYPE AV_TX_FLOAT_RDFT
#define FABS fabsf
#define POW powf
#else
#define SAMPLE_FORMAT double
#define SQRT sqrt
@ -43,6 +47,8 @@
#define ctype AVComplexDouble
#define ftype double
#define TX_TYPE AV_TX_DOUBLE_RDFT
#define FABS fabs
#define POW pow
#endif
#define fn3(a,b) a##_##b
@ -139,95 +145,32 @@ end:
av_free(mag);
}
static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, int ch,
ftype *time)
static ftype fn(ir_gain)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, const ftype *time)
{
ftype ch_gain = 1;
ftype ch_gain, sum = 0;
switch (s->gtype) {
case -1:
if (s->ir_norm < 0.f) {
ch_gain = 1;
break;
case 0:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += FFABS(time[i]);
ch_gain = 1. / sum;
}
break;
case 1:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i];
ch_gain = 1. / sum;
}
break;
case 2:
{
ftype sum = 0;
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i] * time[i];
ch_gain = 1. / SQRT(sum);
}
break;
case 3:
case 4:
{
ftype *inc, *outc, scale, power;
AVTXContext *tx;
av_tx_fn tx_fn;
int ret, size;
size = 1 << av_ceil_log2_c(cur_nb_taps);
inc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
outc = av_calloc(size + 2, sizeof(SAMPLE_FORMAT));
if (!inc || !outc) {
av_free(outc);
av_free(inc);
break;
}
scale = 1.;
ret = av_tx_init(&tx, &tx_fn, TX_TYPE, 0, size, &scale, 0);
if (ret < 0) {
av_free(outc);
av_free(inc);
break;
}
{
memcpy(inc, time, cur_nb_taps * sizeof(SAMPLE_FORMAT));
tx_fn(tx, outc, inc, sizeof(SAMPLE_FORMAT));
power = 0;
if (s->gtype == 3) {
for (int i = 0; i < size / 2 + 1; i++)
power = FFMAX(power, HYPOT(outc[i * 2], outc[i * 2 + 1]));
} else {
ftype sum = 0;
for (int i = 0; i < size / 2 + 1; i++)
sum += HYPOT(outc[i * 2], outc[i * 2 + 1]);
power = SQRT(sum / (size / 2 + 1));
}
ch_gain = 1. / power;
}
av_tx_uninit(&tx);
av_free(outc);
av_free(inc);
}
break;
default:
return AVERROR_BUG;
} else if (s->ir_norm == 0.f) {
for (int i = 0; i < cur_nb_taps; i++)
sum += time[i];
ch_gain = 1. / sum;
} else {
ftype ir_norm = s->ir_norm;
for (int i = 0; i < cur_nb_taps; i++)
sum += POW(FABS(time[i]), ir_norm);
ch_gain = 1. / POW(sum, 1. / ir_norm);
}
return ch_gain;
}
static void fn(ir_scale)(AVFilterContext *ctx, AudioFIRContext *s,
int cur_nb_taps, int ch,
ftype *time, ftype ch_gain)
{
if (ch_gain != 1. || s->ir_gain != 1.) {
ftype gain = ch_gain * s->ir_gain;
@ -238,8 +181,6 @@ static int fn(get_power)(AVFilterContext *ctx, AudioFIRContext *s,
s->fdsp->vector_dmul_scalar(time, time, gain, FFALIGN(cur_nb_taps, 8));
#endif
}
return 0;
}
static void fn(convert_channel)(AVFilterContext *ctx, AudioFIRContext *s, int ch,

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