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avfilter/af_afir: split input frames from impulse response frames

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This commit is contained in:
Paul B Mahol 2020-01-08 19:23:45 +01:00
parent 929e5159bc
commit e364fe4cca
2 changed files with 15 additions and 14 deletions
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26
libavfilter/af_afir.c
View File

@ -59,7 +59,7 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset) static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
{ {
AudioFIRContext *s = ctx->priv; AudioFIRContext *s = ctx->priv;
const float *in = (const float *)s->in[0]->extended_data[ch] + offset; const float *in = (const float *)s->in->extended_data[ch] + offset;
float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset; float *block, *buf, *ptr = (float *)out->extended_data[ch] + offset;
const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset); const int nb_samples = FFMIN(s->min_part_size, out->nb_samples - offset);
int n, i, j; int n, i, j;
@ -175,7 +175,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
if (s->pts == AV_NOPTS_VALUE) if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts; s->pts = in->pts;
s->in[0] = in; s->in = in;
ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels, ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
ff_filter_get_nb_threads(ctx))); ff_filter_get_nb_threads(ctx)));
@ -184,7 +184,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_frame_free(&in); av_frame_free(&in);
s->in[0] = NULL; s->in = NULL;
return ff_filter_frame(outlink, out); return ff_filter_frame(outlink, out);
} }
@ -255,9 +255,9 @@ static void draw_response(AVFilterContext *ctx, AVFrame *out)
if (!mag || !phase || !delay) if (!mag || !phase || !delay)
goto end; goto end;
channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1); channel = av_clip(s->ir_channel, 0, s->ir[0]->channels - 1);
for (i = 0; i < s->w; i++) { for (i = 0; i < s->w; i++) {
const float *src = (const float *)s->in[1]->extended_data[channel]; const float *src = (const float *)s->ir[0]->extended_data[channel];
double w = i * M_PI / (s->w - 1); double w = i * M_PI / (s->w - 1);
double div, real_num = 0., imag_num = 0., real = 0., imag = 0.; double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
@ -404,7 +404,7 @@ static int convert_coeffs(AVFilterContext *ctx)
part_size = FFMIN(part_size, max_part_size); part_size = FFMIN(part_size, max_part_size);
} }
ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]); ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->ir[0]);
if (ret < 0) if (ret < 0)
return ret; return ret;
if (ret == 0) if (ret == 0)
@ -421,7 +421,7 @@ static int convert_coeffs(AVFilterContext *ctx)
break; break;
case 0: case 0:
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
for (i = 0; i < s->nb_taps; i++) for (i = 0; i < s->nb_taps; i++)
power += FFABS(time[i]); power += FFABS(time[i]);
@ -430,7 +430,7 @@ static int convert_coeffs(AVFilterContext *ctx)
break; break;
case 1: case 1:
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
for (i = 0; i < s->nb_taps; i++) for (i = 0; i < s->nb_taps; i++)
power += time[i]; power += time[i];
@ -439,7 +439,7 @@ static int convert_coeffs(AVFilterContext *ctx)
break; break;
case 2: case 2:
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
for (i = 0; i < s->nb_taps; i++) for (i = 0; i < s->nb_taps; i++)
power += time[i] * time[i]; power += time[i] * time[i];
@ -453,7 +453,7 @@ static int convert_coeffs(AVFilterContext *ctx)
s->gain = FFMIN(s->gain * s->ir_gain, 1.f); s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain); av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4)); s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4));
} }
@ -462,7 +462,7 @@ static int convert_coeffs(AVFilterContext *ctx)
av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments); av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; float *time = (float *)s->ir[0]->extended_data[!s->one2many * ch];
int toffset = 0; int toffset = 0;
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
@ -510,7 +510,7 @@ static int convert_coeffs(AVFilterContext *ctx)
} }
} }
av_frame_free(&s->in[1]); av_frame_free(&s->ir[0]);
s->have_coeffs = 1; s->have_coeffs = 1;
return 0; return 0;
@ -727,7 +727,7 @@ static av_cold void uninit(AVFilterContext *ctx)
} }
av_freep(&s->fdsp); av_freep(&s->fdsp);
av_frame_free(&s->in[1]); av_frame_free(&s->ir[0]);
for (int i = 0; i < ctx->nb_outputs; i++) for (int i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name); av_freep(&ctx->output_pads[i].name);

3
libavfilter/af_afir.h
View File

@ -87,7 +87,8 @@ typedef struct AudioFIRContext {
AudioFIRSegment seg[1024]; AudioFIRSegment seg[1024];
int nb_segments; int nb_segments;
AVFrame *in[2]; AVFrame *in;
AVFrame *ir[32];
AVFrame *video; AVFrame *video;
int min_part_size; int min_part_size;
int64_t pts; int64_t pts;