#include <float.h>
-#include "libavutil/audio_fifo.h"
+#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intreadwrite.h"
#include "audio.h"
#include "avfilter.h"
+#include "filters.h"
#include "formats.h"
#include "internal.h"
#include "af_afir.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];
- int index1 = (s->index + 1) % 3;
- int index2 = (s->index + 2) % 3;
- float *sum = s->sum[ch];
- AVFrame *out = arg;
- float *block;
- float *dst;
+ 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->fft_length);
- block = s->block[ch] + s->part_index * s->block_size;
- memset(block, 0, sizeof(*block) * s->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;
+ }
- s->fdsp->vector_fmul_scalar(block + s->part_size, src, s->dry_gain, FFALIGN(s->nb_samples, 4));
- emms_c();
+ 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));
- av_rdft_calc(s->rdft[ch], block);
- block[2 * s->part_size] = block[1];
- block[1] = 0;
+ dst += seg->output_offset[ch];
+ fir_fadd(s, ptr, dst, nb_samples);
+ continue;
+ }
- j = s->part_index;
+ if (seg->part_size < 8) {
+ memset(dst, 0, sizeof(*dst) * seg->part_size * seg->nb_partitions);
- for (i = 0; i < s->nb_partitions; i++) {
- const int coffset = i * s->coeff_size;
- const FFTComplex *coeff = s->coeff[ch * !s->one2many] + coffset;
+ j = seg->part_index[ch];
- block = s->block[ch] + j * s->block_size;
- s->fcmul_add(sum, block, (const float *)coeff, s->part_size);
+ 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;
- if (j == 0)
- j = s->nb_partitions;
- j--;
- }
+ direct(src, coeff, nb_samples, dst);
- sum[1] = sum[2 * s->part_size];
- av_rdft_calc(s->irdft[ch], sum);
+ if (j == 0)
+ j = seg->nb_partitions;
+ j--;
+ }
- dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
- for (n = 0; n < s->part_size; n++) {
- dst[n] += sum[n];
- }
+ 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;
+ }
+
+ 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));
+
+ memcpy(block, src, sizeof(*src) * seg->part_size);
+
+ av_rdft_calc(seg->rdft[ch], block);
+ block[2 * seg->part_size] = block[1];
+ block[1] = 0;
+
+ j = seg->part_index[ch];
+
+ 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;
+
+ 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));
- dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
+ buf = (float *)seg->buffer->extended_data[ch];
+ memcpy(buf, sum + seg->part_size, seg->part_size * sizeof(*buf));
- memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
+ seg->part_index[ch] = (seg->part_index[ch] + 1) % seg->nb_partitions;
- dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
+ memmove(src, src + s->min_part_size, (seg->input_size - s->min_part_size) * sizeof(*src));
- if (out) {
- float *ptr = (float *)out->extended_data[ch];
- s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
+ fir_fadd(s, ptr, dst, nb_samples);
+ }
+
+ 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;
}
return 0;
}
-static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
+static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
- AVFilterContext *ctx = outlink->src;
- AVFrame *out = NULL;
- int ret;
-
- s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
+ AudioFIRContext *s = ctx->priv;
- if (!s->want_skip) {
- out = ff_get_audio_buffer(outlink, s->nb_samples);
- if (!out)
- return AVERROR(ENOMEM);
+ for (int offset = 0; offset < out->nb_samples; offset += s->min_part_size) {
+ fir_quantum(ctx, out, ch, offset);
}
- s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
- if (!s->in[0]) {
- av_frame_free(&out);
- return AVERROR(ENOMEM);
- }
+ return 0;
+}
- av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
+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;
- ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
+ for (int ch = start; ch < end; ch++) {
+ fir_channel(ctx, out, ch);
+ }
- s->part_index = (s->part_index + 1) % s->nb_partitions;
+ return 0;
+}
- av_audio_fifo_drain(s->fifo[0], s->nb_samples);
+static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AVFrame *out = NULL;
- if (!s->want_skip) {
- 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);
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
}
- s->index++;
- if (s->index == 3)
- s->index = 0;
+ if (s->pts == AV_NOPTS_VALUE)
+ s->pts = in->pts;
+ s->in = in;
+ ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels,
+ ff_filter_get_nb_threads(ctx)));
- av_frame_free(&s->in[0]);
+ 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);
- if (s->want_skip == 1) {
- s->want_skip = 0;
- ret = 0;
- } else {
- ret = ff_filter_frame(outlink, out);
- }
+ av_frame_free(&in);
+ s->in = NULL;
- return ret;
+ return ff_filter_frame(outlink, out);
}
static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color)
static void draw_response(AVFilterContext *ctx, AVFrame *out)
{
AudioFIRContext *s = ctx->priv;
- float *mag, *phase, min = FLT_MAX, max = FLT_MIN;
- int prev_ymag = -1, prev_yphase = -1;
+ float *mag, *phase, *delay, min = FLT_MAX, max = FLT_MIN;
+ float min_delay = FLT_MAX, max_delay = FLT_MIN;
+ int prev_ymag = -1, prev_yphase = -1, prev_ydelay = -1;
char text[32];
int channel, i, x;
phase = av_malloc_array(s->w, sizeof(*phase));
mag = av_malloc_array(s->w, sizeof(*mag));
- if (!mag || !phase)
+ delay = av_malloc_array(s->w, sizeof(*delay));
+ 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 real = 0.;
- double imag = 0.;
+ double div, real_num = 0., imag_num = 0., real = 0., imag = 0.;
for (x = 0; x < s->nb_taps; x++) {
real += cos(-x * w) * src[x];
imag += sin(-x * w) * src[x];
+ real_num += cos(-x * w) * src[x] * x;
+ imag_num += sin(-x * w) * src[x] * x;
}
mag[i] = hypot(real, imag);
phase[i] = atan2(imag, real);
+ div = real * real + imag * imag;
+ delay[i] = (real_num * real + imag_num * imag) / div;
min = fminf(min, mag[i]);
max = fmaxf(max, mag[i]);
+ min_delay = fminf(min_delay, delay[i]);
+ max_delay = fmaxf(max_delay, delay[i]);
}
for (i = 0; i < s->w; i++) {
int ymag = mag[i] / max * (s->h - 1);
+ int ydelay = (delay[i] - min_delay) / (max_delay - min_delay) * (s->h - 1);
int yphase = (0.5 * (1. + phase[i] / M_PI)) * (s->h - 1);
ymag = s->h - 1 - av_clip(ymag, 0, s->h - 1);
yphase = s->h - 1 - av_clip(yphase, 0, s->h - 1);
+ ydelay = s->h - 1 - av_clip(ydelay, 0, s->h - 1);
if (prev_ymag < 0)
prev_ymag = ymag;
if (prev_yphase < 0)
prev_yphase = yphase;
+ if (prev_ydelay < 0)
+ prev_ydelay = ydelay;
draw_line(out, i, ymag, FFMAX(i - 1, 0), prev_ymag, 0xFFFF00FF);
draw_line(out, i, yphase, FFMAX(i - 1, 0), prev_yphase, 0xFF00FF00);
+ draw_line(out, i, ydelay, FFMAX(i - 1, 0), prev_ydelay, 0xFF00FFFF);
prev_ymag = ymag;
prev_yphase = yphase;
+ prev_ydelay = ydelay;
}
if (s->w > 400 && s->h > 100) {
drawtext(out, 2, 12, "Min Magnitude:", 0xDDDDDDDD);
snprintf(text, sizeof(text), "%.2f", min);
drawtext(out, 15 * 8 + 2, 12, text, 0xDDDDDDDD);
+
+ drawtext(out, 2, 22, "Max Delay:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", max_delay);
+ drawtext(out, 11 * 8 + 2, 22, text, 0xDDDDDDDD);
+
+ drawtext(out, 2, 32, "Min Delay:", 0xDDDDDDDD);
+ snprintf(text, sizeof(text), "%.2f", min_delay);
+ drawtext(out, 11 * 8 + 2, 32, text, 0xDDDDDDDD);
}
end:
+ av_free(delay);
av_free(phase);
av_free(mag);
}
-static int convert_coeffs(AVFilterContext *ctx)
+static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
+ int offset, int nb_partitions, int part_size)
{
AudioFIRContext *s = ctx->priv;
- int i, ch, n, N;
- s->nb_taps = av_audio_fifo_size(s->fifo[1]);
- if (s->nb_taps <= 0)
- return AVERROR(EINVAL);
+ seg->rdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->rdft));
+ seg->irdft = av_calloc(ctx->inputs[0]->channels, sizeof(*seg->irdft));
+ if (!seg->rdft || !seg->irdft)
+ return AVERROR(ENOMEM);
- for (n = 4; (1 << n) < s->nb_taps; n++);
- N = FFMIN(n, 16);
- s->ir_length = 1 << n;
- s->fft_length = (1 << (N + 1)) + 1;
- s->part_size = 1 << (N - 1);
- s->block_size = FFALIGN(s->fft_length, 32);
- s->coeff_size = FFALIGN(s->part_size + 1, 32);
- s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
- s->nb_coeffs = s->ir_length + s->nb_partitions;
-
- for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
- s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
- if (!s->sum[ch])
- return AVERROR(ENOMEM);
- }
+ 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;
+
+ 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 (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- s->coeff[ch] = av_calloc(s->nb_partitions * s->coeff_size, sizeof(**s->coeff));
- if (!s->coeff[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])
return AVERROR(ENOMEM);
}
- for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
- s->block[ch] = av_calloc(s->nb_partitions * s->block_size, sizeof(**s->block));
- if (!s->block[ch])
- return AVERROR(ENOMEM);
+ 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);
+ 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);
- for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
- s->rdft[ch] = av_rdft_init(N, DFT_R2C);
- s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
- if (!s->rdft[ch] || !s->irdft[ch])
- return AVERROR(ENOMEM);
+ if (seg->irdft) {
+ for (int ch = 0; ch < s->nb_channels; ch++) {
+ av_rdft_end(seg->irdft[ch]);
+ }
}
+ av_freep(&seg->irdft);
- s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
- if (!s->in[1])
- return AVERROR(ENOMEM);
+ av_freep(&seg->output_offset);
+ av_freep(&seg->part_index);
- s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
- if (!s->buffer)
- return AVERROR(ENOMEM);
+ 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, cur_nb_taps;
+ float power = 0;
- av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
+ if (!s->nb_taps) {
+ int part_size, max_part_size;
+ int left, offset = 0;
+
+ s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]);
+ if (s->nb_taps <= 0)
+ return AVERROR(EINVAL);
+
+ if (s->minp > s->maxp) {
+ s->maxp = s->minp;
+ }
+
+ 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);
- if (s->again) {
- float power = 0;
+ s->gain = 1;
+ cur_nb_taps = s->ir[s->selir]->nb_samples;
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ switch (s->gtype) {
+ case -1:
+ /* nothing to do */
+ break;
+ case 0:
+ 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 + s->selir]->channels / power;
+ break;
+ case 1:
+ 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 < cur_nb_taps; i++)
+ power += time[i];
+ }
+ s->gain = ctx->inputs[1 + s->selir]->channels / power;
+ break;
+ case 2:
+ 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 < cur_nb_taps; i++)
+ power += time[i] * time[i];
+ }
+ s->gain = sqrtf(ch / power);
+ break;
+ default:
+ return AVERROR_BUG;
+ }
- s->gain = sqrtf(1.f / (ctx->inputs[1]->channels * power)) / (sqrtf(ctx->inputs[1]->channels));
- for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
- float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
+ 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 + 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 = s->block[ch];
- FFTComplex *coeff = s->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->nb_partitions; i++) {
- const float scale = 1.f / s->part_size;
- const int toffset = i * s->part_size;
- const int coffset = i * s->coeff_size;
- const int boffset = s->part_size;
- const int remaining = s->nb_taps - (i * s->part_size);
- const int size = remaining >= s->part_size ? s->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];
- memset(block, 0, sizeof(*block) * s->fft_length);
- memcpy(block + boffset, time + toffset, size * sizeof(*block));
+ av_log(ctx, AV_LOG_DEBUG, "segment: %d\n", segment);
- av_rdft_calc(s->rdft[0], block);
+ 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;
- coeff[coffset].re = block[0] * scale;
- coeff[coffset].im = 0;
- for (n = 1; n < s->part_size; n++) {
- coeff[coffset + n].re = block[2 * n] * scale;
- coeff[coffset + n].im = block[2 * n + 1] * scale;
+ if (size < 8) {
+ for (n = 0; n < size; n++)
+ coeff[coffset + n].re = time[toffset + n];
+
+ toffset += size;
+ continue;
+ }
+
+ memset(block, 0, sizeof(*block) * seg->fft_length);
+ memcpy(block, time + toffset, size * sizeof(*block));
+
+ av_rdft_calc(seg->rdft[0], block);
+
+ 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->part_size].re = block[1] * scale;
- coeff[coffset + s->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->nb_partitions);
- av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
- av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
-
s->have_coeffs = 1;
return 0;
}
-static int read_ir(AVFilterLink *link, AVFrame *frame)
+static int check_ir(AVFilterLink *link)
{
AVFilterContext *ctx = link->dst;
AudioFIRContext *s = ctx->priv;
- int nb_taps, max_nb_taps, ret;
+ int nb_taps, max_nb_taps;
- ret = av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
- frame->nb_samples);
- av_frame_free(&frame);
- if (ret < 0)
- return ret;
-
- nb_taps = av_audio_fifo_size(s->fifo[1]);
+ nb_taps = ff_inlink_queued_samples(link);
max_nb_taps = s->max_ir_len * ctx->outputs[0]->sample_rate;
if (nb_taps > max_nb_taps) {
av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
return 0;
}
-static int filter_frame(AVFilterLink *link, AVFrame *frame)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = link->dst;
AudioFIRContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
- int ret;
+ int ret, status, available, wanted;
+ AVFrame *in = NULL;
+ int64_t pts;
- ret = av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
- frame->nb_samples);
- if (ret > 0 && s->pts == AV_NOPTS_VALUE)
- s->pts = frame->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[s->selir]) {
+ ret = check_ir(ctx->inputs[1 + s->selir]);
+ if (ret < 0)
+ return ret;
- av_frame_free(&frame);
+ if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF)
+ s->eof_coeffs[s->selir] = 1;
- if (ret < 0)
- return ret;
+ if (!s->eof_coeffs[s->selir]) {
+ if (ff_outlink_frame_wanted(ctx->outputs[0]))
+ 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 + 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;
}
- if (s->response && s->have_coeffs) {
- s->video->pts = s->pts;
- ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
- if (ret < 0)
- return ret;
- }
+ 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);
- if (s->have_coeffs) {
- while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
- ret = fir_frame(s, outlink);
- if (ret < 0)
- return ret;
+ if (ret < 0)
+ return ret;
+
+ if (s->response && s->have_coeffs) {
+ int64_t old_pts = s->video->pts;
+ 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;
+ clone = av_frame_clone(s->video);
+ if (!clone)
+ return AVERROR(ENOMEM);
+ return ff_filter_frame(ctx->outputs[1], clone);
}
}
- return 0;
-}
-static int request_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- AudioFIRContext *s = ctx->priv;
- int ret;
+ if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) {
+ ff_filter_set_ready(ctx, 10);
+ return 0;
+ }
- if (!s->eof_coeffs) {
- ret = ff_request_frame(ctx->inputs[1]);
- if (ret == AVERROR_EOF) {
- s->eof_coeffs = 1;
- ret = 0;
+ if (ff_inlink_acknowledge_status(ctx->inputs[0], &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ ff_outlink_set_status(ctx->outputs[0], status, pts);
+ if (s->response)
+ ff_outlink_set_status(ctx->outputs[1], status, pts);
+ return 0;
}
- return ret;
}
- ret = ff_request_frame(ctx->inputs[0]);
- if (ret == AVERROR_EOF && s->have_coeffs) {
- if (s->need_padding) {
- AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
- if (!silence)
- return AVERROR(ENOMEM);
- ret = av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
- silence->nb_samples);
- av_frame_free(&silence);
- if (ret < 0)
- return ret;
- s->need_padding = 0;
- }
+ if (ff_outlink_frame_wanted(ctx->outputs[0]) &&
+ !ff_outlink_get_status(ctx->inputs[0])) {
+ ff_inlink_request_frame(ctx->inputs[0]);
+ return 0;
+ }
- while (av_audio_fifo_size(s->fifo[0]) > 0) {
- ret = fir_frame(s, outlink);
- if (ret < 0)
- return ret;
- }
- ret = AVERROR_EOF;
+ if (s->response &&
+ ff_outlink_frame_wanted(ctx->outputs[1]) &&
+ !ff_outlink_get_status(ctx->inputs[0])) {
+ ff_inlink_request_frame(ctx->inputs[0]);
+ return 0;
}
- return ret;
+
+ return FFERROR_NOT_READY;
}
static int query_formats(AVFilterContext *ctx)
AV_PIX_FMT_RGB0,
AV_PIX_FMT_NONE
};
- int ret, i;
+ int ret;
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;
}
layouts = ff_all_channel_counts();
- if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
- return ret;
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
+ if (s->ir_format) {
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+ } else {
+ AVFilterChannelLayouts *mono = NULL;
+
+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts)) < 0)
+ return ret;
+ if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0)
+ return ret;
- for (i = 0; i < 2; i++) {
- layouts = ff_all_channel_counts();
- if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->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;
- if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
- ctx->inputs[1]->channels != 1) {
- av_log(ctx, AV_LOG_ERROR,
- "Second input must have same number of channels as first input or "
- "exactly 1 channel.\n");
- return AVERROR(EINVAL);
- }
-
- 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->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
- s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
- if (!s->fifo[0] || !s->fifo[1])
- return AVERROR(ENOMEM);
-
- s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
- s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
- s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
- s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
- s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
- if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
- return AVERROR(ENOMEM);
-
s->nb_channels = outlink->channels;
- s->nb_coef_channels = ctx->inputs[1]->channels;
- s->want_skip = 1;
- s->need_padding = 1;
+ s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels;
s->pts = AV_NOPTS_VALUE;
return 0;
static av_cold void uninit(AVFilterContext *ctx)
{
AudioFIRContext *s = ctx->priv;
- int ch;
-
- if (s->sum) {
- for (ch = 0; ch < s->nb_channels; ch++) {
- av_freep(&s->sum[ch]);
- }
- }
- av_freep(&s->sum);
-
- if (s->coeff) {
- for (ch = 0; ch < s->nb_coef_channels; ch++) {
- av_freep(&s->coeff[ch]);
- }
- }
- av_freep(&s->coeff);
- if (s->block) {
- for (ch = 0; ch < s->nb_channels; ch++) {
- av_freep(&s->block[ch]);
- }
+ for (int i = 0; i < s->nb_segments; i++) {
+ uninit_segment(ctx, &s->seg[i]);
}
- av_freep(&s->block);
- if (s->rdft) {
- for (ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(s->rdft[ch]);
- }
- }
- av_freep(&s->rdft);
+ av_freep(&s->fdsp);
- if (s->irdft) {
- for (ch = 0; ch < s->nb_channels; ch++) {
- av_rdft_end(s->irdft[ch]);
- }
+ for (int i = 0; i < s->nb_irs; i++) {
+ av_frame_free(&s->ir[i]);
}
- av_freep(&s->irdft);
- av_frame_free(&s->in[0]);
- av_frame_free(&s->in[1]);
- av_frame_free(&s->buffer);
+ for (unsigned i = 1; i < ctx->nb_inputs; i++)
+ av_freep(&ctx->input_pads[i].name);
- av_audio_fifo_free(s->fifo[0]);
- av_audio_fifo_free(s->fifo[1]);
-
- av_freep(&s->fdsp);
-
- for (int i = 0; i < ctx->nb_outputs; i++)
- av_freep(&ctx->output_pads[i].name);
av_frame_free(&s->video);
}
outlink->sample_aspect_ratio = (AVRational){1,1};
outlink->w = s->w;
outlink->h = s->h;
+ outlink->frame_rate = s->frame_rate;
+ outlink->time_base = av_inv_q(outlink->frame_rate);
av_frame_free(&s->video);
s->video = ff_get_video_buffer(outlink, outlink->w, outlink->h);
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,
- .request_frame = request_frame,
+ 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,
- .filter_frame = filter_frame,
- },{
- .name = "ir",
- .type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = read_ir,
- },
- { 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)
{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AF },
{ "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
- { "again", "enable auto gain", OFFSET(again), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
+ { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 2, 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" },
+ { "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" },
+ { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "irfmt" },
{ "maxir", "set max IR length", OFFSET(max_ir_len), AV_OPT_TYPE_FLOAT, {.dbl=30}, 0.1, 60, AF },
{ "response", "show IR frequency response", OFFSET(response), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, VF },
{ "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=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,
};