#include <float.h>
+#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
sum[2 * n] += t[2 * n] * c[2 * n];
}
-static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
+static void direct(const float *in, const FFTComplex *ir, int len, float *out)
+{
+ for (int n = 0; n < len; n++)
+ for (int m = 0; m <= n; m++)
+ out[n] += ir[m].re * in[n - m];
+}
+
+static void fir_fadd(AudioFIRContext *s, float *dst, const float *src, int nb_samples)
+{
+ if ((nb_samples & 15) == 0 && nb_samples >= 16) {
+ s->fdsp->vector_fmac_scalar(dst, src, 1.f, nb_samples);
+ } else {
+ for (int n = 0; n < nb_samples; n++)
+ dst[n] += src[n];
+ }
+}
+
+static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset)
{
AudioFIRContext *s = ctx->priv;
- const float *src = (const float *)s->in[0]->extended_data[ch];
- float *sum = (float *)s->seg.sum->extended_data[ch];
- AVFrame *out = arg;
- float *block, *dst, *ptr;
+ const float *in = (const float *)s->in->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);
int n, i, j;
- memset(sum, 0, sizeof(*sum) * s->seg.fft_length);
- block = (float *)s->seg.block->extended_data[ch] + s->seg.part_index * s->seg.block_size;
- memset(block, 0, sizeof(*block) * s->seg.fft_length);
+ for (int segment = 0; segment < s->nb_segments; segment++) {
+ AudioFIRSegment *seg = &s->seg[segment];
+ float *src = (float *)seg->input->extended_data[ch];
+ float *dst = (float *)seg->output->extended_data[ch];
+ float *sum = (float *)seg->sum->extended_data[ch];
+
+ if (s->min_part_size >= 8) {
+ s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4));
+ emms_c();
+ } else {
+ for (n = 0; n < nb_samples; n++)
+ src[seg->input_offset + n] = in[n] * s->dry_gain;
+ }
+
+ seg->output_offset[ch] += s->min_part_size;
+ if (seg->output_offset[ch] == seg->part_size) {
+ seg->output_offset[ch] = 0;
+ } else {
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ dst += seg->output_offset[ch];
+ fir_fadd(s, ptr, dst, nb_samples);
+ continue;
+ }
+
+ if (seg->part_size < 8) {
+ memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
+
+ j = seg->part_index[ch];
+
+ for (i = 0; i < seg->nb_partitions; i++) {
+ const int coffset = j * seg->coeff_size;
+ const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
+
+ direct(src, coeff, nb_samples, dst);
+
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
+
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ for (n = 0; n < nb_samples; n++) {
+ ptr[n] += dst[n];
+ }
+ continue;
+ }
- s->fdsp->vector_fmul_scalar(block, src, s->dry_gain, FFALIGN(out->nb_samples, 4));
- emms_c();
+ memset(sum, 0, sizeof(*sum) * seg->fft_length);
+ block = (float *)seg->block->extended_data[ch] + seg->part_index[ch] * seg->block_size;
+ memset(block + seg->part_size, 0, sizeof(*block) * (seg->fft_length - seg->part_size));
- av_rdft_calc(s->seg.rdft[ch], block);
- block[2 * s->seg.part_size] = block[1];
- block[1] = 0;
+ memcpy(block, src, sizeof(*src) * seg->part_size);
- j = s->seg.part_index;
+ av_rdft_calc(seg->rdft[ch], block);
+ block[2 * seg->part_size] = block[1];
+ block[1] = 0;
- for (i = 0; i < s->seg.nb_partitions; i++) {
- const int coffset = i * s->seg.coeff_size;
- const float *block = (const float *)s->seg.block->extended_data[ch] + j * s->seg.block_size;
- const FFTComplex *coeff = s->seg.coeff[ch * !s->one2many] + coffset;
+ j = seg->part_index[ch];
- s->fcmul_add(sum, block, (const float *)coeff, s->seg.part_size);
+ for (i = 0; i < seg->nb_partitions; i++) {
+ const int coffset = j * seg->coeff_size;
+ const float *block = (const float *)seg->block->extended_data[ch] + i * seg->block_size;
+ const FFTComplex *coeff = (const FFTComplex *)seg->coeff->extended_data[ch * !s->one2many] + coffset;
- if (j == 0)
- j = s->seg.nb_partitions;
- j--;
+ s->afirdsp.fcmul_add(sum, block, (const float *)coeff, seg->part_size);
+
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
+
+ sum[1] = sum[2 * seg->part_size];
+ av_rdft_calc(seg->irdft[ch], sum);
+
+ buf = (float *)seg->buffer->extended_data[ch];
+ fir_fadd(s, buf, sum, seg->part_size);
+
+ memcpy(dst, buf, seg->part_size * sizeof(*dst));
+
+ buf = (float *)seg->buffer->extended_data[ch];
+ memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
+
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
+
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
+
+ fir_fadd(s, ptr, dst, nb_samples);
}
- sum[1] = sum[2 * s->seg.part_size];
- av_rdft_calc(s->seg.irdft[ch], sum);
+ if (s->min_part_size >= 8) {
+ s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4));
+ emms_c();
+ } else {
+ for (n = 0; n < nb_samples; n++)
+ ptr[n] *= s->wet_gain;
+ }
- dst = (float *)s->seg.buffer->extended_data[ch];
- for (n = 0; n < s->seg.part_size; n++) {
- dst[n] += sum[n];
+ return 0;
+}
+
+static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
+{
+ AudioFIRContext *s = ctx->priv;
+
+ for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
+ fir_quantum(ctx, out, ch, offset);
}
- ptr = (float *)out->extended_data[ch];
- s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
- emms_c();
+ return 0;
+}
- dst = (float *)s->seg.buffer->extended_data[ch];
- memcpy(dst, sum + s->seg.part_size, s->seg.part_size * sizeof(*dst));
+static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AVFrame *out = arg;
+ const int start = (out->channels * jobnr) / nb_jobs;
+ const int end = (out->channels * (jobnr+1)) / nb_jobs;
+
+ for (int ch = start; ch < end; ch++) {
+ fir_channel(ctx, out, ch);
+ }
return 0;
}
if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
- s->in[0] = in;
- ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
-
- s->seg.part_index = (s->seg.part_index + 1) % s->seg.nb_partitions;
+ s->in = in;
+ ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
+ ff_filter_get_nb_threads(ctx)));
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_frame_free(&in);
- s->in[0] = NULL;
+ s->in = NULL;
return ff_filter_frame(outlink, out);
}
if (!mag || !phase || !delay)
goto end;
- channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1);
+ channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1);
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[s->selir]->extended_data[channel];
double w = i * M_PI / (s->w - 1);
double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
av_free(mag);
}
-static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int nb_partitions, int part_size)
+static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
+ int offset, int nb_partitions, int part_size)
{
- seg->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*seg->coeff));
+ AudioFIRContext *s = ctx->priv;
+
seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
- if (!seg->coeff || !seg->rdft || !seg->irdft)
+ if (!seg->rdft || !seg->irdft)
return AVERROR(ENOMEM);
- seg->fft_length = part_size * 4 + 1;
- seg->part_size = part_size;
- seg->block_size = FFALIGN(seg->fft_length, 32);
- seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
+ seg->fft_length = part_size * 2 + 1;
+ seg->part_size = part_size;
+ seg->block_size = FFALIGN(seg->fft_length, 32);
+ seg->coeff_size = FFALIGN(seg->part_size + 1, 32);
seg->nb_partitions = nb_partitions;
+ seg->input_size = offset + s->min_part_size;
+ seg->input_offset = offset;
- for (int ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- seg->coeff[ch] = av_calloc(seg->nb_partitions * seg->coeff_size, sizeof(**seg->coeff));
- if (!seg->coeff[ch])
- return AVERROR(ENOMEM);
- }
+ seg->part_index = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->part_index));
+ seg->output_offset = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->output_offset));
+ if (!seg->part_index || !seg->output_offset)
+ return AVERROR(ENOMEM);
- for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) {
+ for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) {
seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C);
seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R);
if (!seg->rdft[ch] || !seg->irdft[ch])
seg->sum = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length);
seg->block = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->block_size);
seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
- if (!seg->buffer || !seg->sum || !seg->block)
+ seg->coeff = ff_get_audio_buffer(ctx->inputs[1 + s->selir], seg->nb_partitions * seg->coeff_size * 2);
+ seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size);
+ seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size);
+ if (!seg->buffer || !seg->sum || !seg->block || !seg->coeff || !seg->input || !seg->output)
return AVERROR(ENOMEM);
return 0;
}
+static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
+{
+ AudioFIRContext *s = ctx->priv;
+
+ if (seg->rdft) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ av_rdft_end(seg->rdft[ch]);
+ }
+ }
+ av_freep(&seg->rdft);
+
+ if (seg->irdft) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ av_rdft_end(seg->irdft[ch]);
+ }
+ }
+ av_freep(&seg->irdft);
+
+ av_freep(&seg->output_offset);
+ av_freep(&seg->part_index);
+
+ av_frame_free(&seg->block);
+ av_frame_free(&seg->sum);
+ av_frame_free(&seg->buffer);
+ av_frame_free(&seg->coeff);
+ av_frame_free(&seg->input);
+ av_frame_free(&seg->output);
+ seg->input_size = 0;
+}
+
static int convert_coeffs(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- int ret, i, ch, n, N;
+ int ret, i, ch, n, cur_nb_taps;
float power = 0;
- s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]);
- if (s->nb_taps <= 0)
- return AVERROR(EINVAL);
+ if (!s->nb_taps) {
+ int part_size, max_part_size;
+ int left, offset = 0;
- for (n = av_log2(s->minp); (1 << n) < s->nb_taps; n++);
- N = FFMIN(n, av_log2(s->maxp));
+ s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
+ if (s->nb_taps <= 0)
+ return AVERROR(EINVAL);
- ret = init_segment(ctx, &s->seg, (s->nb_taps + (1 << N) - 1) / (1 << N), 1 << N);
- if (ret < 0)
- return ret;
+ if (s->minp > s->maxp) {
+ s->maxp = s->minp;
+ }
- ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]);
- if (ret < 0)
- return ret;
- if (ret == 0)
- return AVERROR_BUG;
+ left = s->nb_taps;
+ part_size = 1 << av_log2(s->minp);
+ max_part_size = 1 << av_log2(s->maxp);
+
+ s->min_part_size = part_size;
+
+ for (i = 0; left > 0; i++) {
+ int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
+ int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
+
+ s->nb_segments = i + 1;
+ ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, part_size);
+ if (ret < 0)
+ return ret;
+ offset += nb_partitions * part_size;
+ left -= nb_partitions * part_size;
+ part_size *= 2;
+ part_size = FFMIN(part_size, max_part_size);
+ }
+ }
+
+ if (!s->ir[s->selir]) {
+ ret = ff_inlink_consume_samples(ctx->inputs[1 + s->selir], s->nb_taps, s->nb_taps, &s->ir[s->selir]);
+ if (ret < 0)
+ return ret;
+ if (ret == 0)
+ return AVERROR_BUG;
+ }
if (s->response)
draw_response(ctx, s->video);
s->gain = 1;
+ cur_nb_taps = s->ir[s->selir]->nb_samples;
switch (s->gtype) {
case -1:
/* nothing to do */
break;
case 0:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += FFABS(time[i]);
}
- s->gain = ctx->inputs[1]->channels / power;
+ s->gain = ctx->inputs[1 + s->selir]->channels / power;
break;
case 1:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += time[i];
}
- s->gain = ctx->inputs[1]->channels / power;
+ s->gain = ctx->inputs[1 + s->selir]->channels / power;
break;
case 2:
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
- for (i = 0; i < s->nb_taps; i++)
+ for (i = 0; i < cur_nb_taps; i++)
power += time[i] * time[i];
}
s->gain = sqrtf(ch / power);
s->gain = FFMIN(s->gain * s->ir_gain, 1.f);
av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain);
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->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(cur_nb_taps, 4));
}
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
- float *block = (float *)s->seg.block->extended_data[ch];
- FFTComplex *coeff = s->seg.coeff[ch];
+ av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", cur_nb_taps);
+ av_log(ctx, AV_LOG_DEBUG, "nb_segments: %d\n", s->nb_segments);
+
+ for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) {
+ float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch];
+ int toffset = 0;
for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++)
time[i] = 0;
- for (i = 0; i < s->seg.nb_partitions; i++) {
- const float scale = 1.f / s->seg.part_size;
- const int toffset = i * s->seg.part_size;
- const int coffset = i * s->seg.coeff_size;
- const int remaining = s->nb_taps - (i * s->seg.part_size);
- const int size = remaining >= s->seg.part_size ? s->seg.part_size : remaining;
+ av_log(ctx, AV_LOG_DEBUG, "channel: %d\n", ch);
+
+ for (int segment = 0; segment < s->nb_segments; segment++) {
+ AudioFIRSegment *seg = &s->seg[segment];
+ float *block = (float *)seg->block->extended_data[ch];
+ FFTComplex *coeff = (FFTComplex *)seg->coeff->extended_data[ch];
+
+ av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
+
+ for (i = 0; i < seg->nb_partitions; i++) {
+ const float scale = 1.f / seg->part_size;
+ const int coffset = i * seg->coeff_size;
+ const int remaining = s->nb_taps - toffset;
+ const int size = remaining >= seg->part_size ? seg->part_size : remaining;
+
+ if (size < 8) {
+ for (n = 0; n < size; n++)
+ coeff[coffset + n].re = time[toffset + n];
+
+ toffset += size;
+ continue;
+ }
- memset(block, 0, sizeof(*block) * s->seg.fft_length);
- memcpy(block, time + toffset, size * sizeof(*block));
+ memset(block, 0, sizeof(*block) * seg->fft_length);
+ memcpy(block, time + toffset, size * sizeof(*block));
- av_rdft_calc(s->seg.rdft[0], block);
+ av_rdft_calc(seg->rdft[0], block);
- coeff[coffset].re = block[0] * scale;
- coeff[coffset].im = 0;
- for (n = 1; n < s->seg.part_size; n++) {
- coeff[coffset + n].re = block[2 * n] * scale;
- coeff[coffset + n].im = block[2 * n + 1] * scale;
+ coeff[coffset].re = block[0] * scale;
+ coeff[coffset].im = 0;
+ for (n = 1; n < seg->part_size; n++) {
+ coeff[coffset + n].re = block[2 * n] * scale;
+ coeff[coffset + n].im = block[2 * n + 1] * scale;
+ }
+ coeff[coffset + seg->part_size].re = block[1] * scale;
+ coeff[coffset + seg->part_size].im = 0;
+
+ toffset += size;
}
- coeff[coffset + s->seg.part_size].re = block[1] * scale;
- coeff[coffset + s->seg.part_size].im = 0;
+
+ av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", seg->nb_partitions);
+ av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", seg->part_size);
+ av_log(ctx, AV_LOG_DEBUG, "block size: %d\n", seg->block_size);
+ av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", seg->fft_length);
+ av_log(ctx, AV_LOG_DEBUG, "coeff_size: %d\n", seg->coeff_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_size: %d\n", seg->input_size);
+ av_log(ctx, AV_LOG_DEBUG, "input_offset: %d\n", seg->input_offset);
}
}
- av_frame_free(&s->in[1]);
- av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
- av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->seg.nb_partitions);
- av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->seg.part_size);
- av_log(ctx, AV_LOG_DEBUG, "fft_length: %d\n", s->seg.fft_length);
-
s->have_coeffs = 1;
return 0;
}
-static int check_ir(AVFilterLink *link, AVFrame *frame)
+static int check_ir(AVFilterLink *link)
{
AVFilterContext *ctx = link->dst;
AudioFIRContext *s = ctx->priv;
{
AudioFIRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
+ int ret, status, available, wanted;
AVFrame *in = NULL;
- int ret, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if (s->response)
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx);
- if (!s->eof_coeffs) {
- AVFrame *ir = NULL;
-
- ret = check_ir(ctx->inputs[1], ir);
+ if (!s->eof_coeffs[s->selir]) {
+ ret = check_ir(ctx->inputs[1 + s->selir]);
if (ret < 0)
return ret;
- if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF)
- s->eof_coeffs = 1;
+ if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
+ s->eof_coeffs[s->selir] = 1;
- if (!s->eof_coeffs) {
+ if (!s->eof_coeffs[s->selir]) {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
- ff_inlink_request_frame(ctx->inputs[1]);
+ ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1]))
- ff_inlink_request_frame(ctx->inputs[1]);
+ ff_inlink_request_frame(ctx->inputs[1 + s->selir]);
return 0;
}
}
- if (!s->have_coeffs && s->eof_coeffs) {
+ if (!s->have_coeffs && s->eof_coeffs[s->selir]) {
ret = convert_coeffs(ctx);
if (ret < 0)
return ret;
}
- ret = ff_inlink_consume_samples(ctx->inputs[0], s->seg.part_size, s->seg.part_size, &in);
+ available = ff_inlink_queued_samples(ctx->inputs[0]);
+ wanted = FFMAX(s->min_part_size, (available / s->min_part_size) * s->min_part_size);
+ ret = ff_inlink_consume_samples(ctx->inputs[0], wanted, wanted, &in);
if (ret > 0)
ret = fir_frame(s, in, outlink);
int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base);
if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) {
+ AVFrame *clone;
s->video->pts = new_pts;
- return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
+ clone = av_frame_clone(s->video);
+ if (!clone)
+ return AVERROR(ENOMEM);
+ return ff_filter_frame(ctx->outputs[1], clone);
}
}
- if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->seg.part_size) {
+ if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (s->response) {
AVFilterLink *videolink = ctx->outputs[1];
formats = ff_make_format_list(pix_fmts);
- if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
+ if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
return ret;
}
} else {
AVFilterChannelLayouts *mono = NULL;
- ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
- if (ret)
+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
return ret;
-
- if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0)
- return ret;
- if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
return ret;
- if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0)
+
+ ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO);
+ if (ret)
return ret;
+ for (int i = 1; i < ctx->nb_inputs; i++) {
+ if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0)
+ return ret;
+ }
}
formats = ff_make_format_list(sample_fmts);
AVFilterContext *ctx = outlink->src;
AudioFIRContext *s = ctx->priv;
- s->one2many = ctx->inputs[1]->channels == 1;
+ s->one2many = ctx->inputs[1 + s->selir]->channels == 1;
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
s->nb_channels = outlink->channels;
- s->nb_coef_channels = ctx->inputs[1]->channels;
+ s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
s->pts = AV_NOPTS_VALUE;
return 0;
}
-static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
+static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- if (seg->coeff) {
- for (int ch = 0; ch < s->nb_coef_channels; ch++) {
- av_freep(&seg->coeff[ch]);
- }
+ for (int i = 0; i < s->nb_segments; i++) {
+ uninit_segment(ctx, &s->seg[i]);
}
- av_freep(&seg->coeff);
- if (seg->rdft) {
- for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->rdft[ch]);
- }
- }
- av_freep(&seg->rdft);
+ av_freep(&s->fdsp);
- if (seg->irdft) {
- for (int ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(seg->irdft[ch]);
- }
+ for (int i = 0; i < s->nb_irs; i++) {
+ av_frame_free(&s->ir[i]);
}
- av_freep(&seg->irdft);
- av_frame_free(&seg->block);
- av_frame_free(&seg->sum);
- av_frame_free(&seg->buffer);
-}
-
-static av_cold void uninit(AVFilterContext *ctx)
-{
- AudioFIRContext *s = ctx->priv;
+ for (unsigned i = 1; i < ctx->nb_inputs; i++)
+ av_freep(&ctx->input_pads[i].name);
- uninit_segment(ctx, &s->seg);
-
- av_freep(&s->fdsp);
- av_frame_free(&s->in[1]);
-
- for (int i = 0; i < ctx->nb_outputs; i++)
- av_freep(&ctx->output_pads[i].name);
av_frame_free(&s->video);
}
return 0;
}
+void ff_afir_init(AudioFIRDSPContext *dsp)
+{
+ dsp->fcmul_add = fcmul_add_c;
+
+ if (ARCH_X86)
+ ff_afir_init_x86(dsp);
+}
+
static av_cold int init(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
- pad = (AVFilterPad){
- .name = av_strdup("default"),
- .type = AVMEDIA_TYPE_AUDIO,
- .config_props = config_output,
+ pad = (AVFilterPad) {
+ .name = "main",
+ .type = AVMEDIA_TYPE_AUDIO,
};
- if (!pad.name)
- return AVERROR(ENOMEM);
+ ret = ff_insert_inpad(ctx, 0, &pad);
+ if (ret < 0)
+ return ret;
- if (s->response) {
- vpad = (AVFilterPad){
- .name = av_strdup("filter_response"),
- .type = AVMEDIA_TYPE_VIDEO,
- .config_props = config_video,
+ for (int n = 0; n < s->nb_irs; n++) {
+ pad = (AVFilterPad) {
+ .name = av_asprintf("ir%d", n),
+ .type = AVMEDIA_TYPE_AUDIO,
};
- if (!vpad.name)
+
+ if (!pad.name)
return AVERROR(ENOMEM);
+
+ ret = ff_insert_inpad(ctx, n + 1, &pad);
+ if (ret < 0) {
+ av_freep(&pad.name);
+ return ret;
+ }
}
+ pad = (AVFilterPad) {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ };
+
ret = ff_insert_outpad(ctx, 0, &pad);
- if (ret < 0) {
- av_freep(&pad.name);
+ if (ret < 0)
return ret;
- }
if (s->response) {
+ vpad = (AVFilterPad){
+ .name = "filter_response",
+ .type = AVMEDIA_TYPE_VIDEO,
+ .config_props = config_video,
+ };
+
ret = ff_insert_outpad(ctx, 1, &vpad);
- if (ret < 0) {
- av_freep(&vpad.name);
+ if (ret < 0)
return ret;
- }
}
- s->fcmul_add = fcmul_add_c;
-
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
- if (ARCH_X86)
- ff_afir_init_x86(s);
+ ff_afir_init(&s->afirdsp);
return 0;
}
-static const AVFilterPad afir_inputs[] = {
- {
- .name = "main",
- .type = AVMEDIA_TYPE_AUDIO,
- },{
- .name = "ir",
- .type = AVMEDIA_TYPE_AUDIO,
- },
- { NULL }
-};
+static int process_command(AVFilterContext *ctx,
+ const char *cmd,
+ const char *arg,
+ char *res,
+ int res_len,
+ int flags)
+{
+ AudioFIRContext *s = ctx->priv;
+ int prev_ir = s->selir;
+ int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags);
+
+ if (ret < 0)
+ return ret;
+
+ s->selir = FFMIN(s->nb_irs - 1, s->selir);
+
+ if (prev_ir != s->selir) {
+ s->have_coeffs = 0;
+ }
+
+ return 0;
+}
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OFFSET(x) offsetof(AudioFIRContext, x)
{ "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF },
{ "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF },
- { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=16}, 16, 32768, AF },
- { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 16, 32768, AF },
+ { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF },
+ { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF },
+ { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF },
+ { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR },
{ NULL }
};
AVFILTER_DEFINE_CLASS(afir);
-AVFilter ff_af_afir = {
+const AVFilter ff_af_afir = {
.name = "afir",
- .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
+ .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in additional stream(s)."),
.priv_size = sizeof(AudioFIRContext),
.priv_class = &afir_class,
.query_formats = query_formats,
.init = init,
.activate = activate,
.uninit = uninit,
- .inputs = afir_inputs,
- .flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_DYNAMIC_INPUTS |
+ AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
projects / ffmpeg / blobdiff