| /* |
| * Copyright (c) 2017 Paul B Mahol |
| * |
| * 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 |
| * An arbitrary audio FIR filter |
| */ |
| |
| #include <float.h> |
| |
| #include "libavutil/avstring.h" |
| #include "libavutil/common.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/intreadwrite.h" |
| #include "libavutil/opt.h" |
| #include "libavutil/xga_font_data.h" |
| #include "libavcodec/avfft.h" |
| |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "filters.h" |
| #include "formats.h" |
| #include "internal.h" |
| #include "af_afir.h" |
| |
| static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t len) |
| { |
| int n; |
| |
| for (n = 0; n < len; n++) { |
| const float cre = c[2 * n ]; |
| const float cim = c[2 * n + 1]; |
| const float tre = t[2 * n ]; |
| const float tim = t[2 * n + 1]; |
| |
| sum[2 * n ] += tre * cre - tim * cim; |
| sum[2 * n + 1] += tre * cim + tim * cre; |
| } |
| |
| sum[2 * n] += t[2 * n] * c[2 * n]; |
| } |
| |
| 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 *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; |
| |
| 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; |
| } |
| |
| 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)); |
| |
| 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); |
| } |
| |
| 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_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); |
| } |
| |
| return 0; |
| } |
| |
| 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; |
| } |
| |
| static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AVFrame *out = NULL; |
| |
| out = ff_get_audio_buffer(outlink, in->nb_samples); |
| if (!out) { |
| av_frame_free(&in); |
| return AVERROR(ENOMEM); |
| } |
| |
| 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))); |
| |
| 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 = NULL; |
| |
| return ff_filter_frame(outlink, out); |
| } |
| |
| static void drawtext(AVFrame *pic, int x, int y, const char *txt, uint32_t color) |
| { |
| const uint8_t *font; |
| int font_height; |
| int i; |
| |
| font = avpriv_cga_font, font_height = 8; |
| |
| for (i = 0; txt[i]; i++) { |
| int char_y, mask; |
| |
| uint8_t *p = pic->data[0] + y * pic->linesize[0] + (x + i * 8) * 4; |
| for (char_y = 0; char_y < font_height; char_y++) { |
| for (mask = 0x80; mask; mask >>= 1) { |
| if (font[txt[i] * font_height + char_y] & mask) |
| AV_WL32(p, color); |
| p += 4; |
| } |
| p += pic->linesize[0] - 8 * 4; |
| } |
| } |
| } |
| |
| static void draw_line(AVFrame *out, int x0, int y0, int x1, int y1, uint32_t color) |
| { |
| int dx = FFABS(x1-x0); |
| int dy = FFABS(y1-y0), sy = y0 < y1 ? 1 : -1; |
| int err = (dx>dy ? dx : -dy) / 2, e2; |
| |
| for (;;) { |
| AV_WL32(out->data[0] + y0 * out->linesize[0] + x0 * 4, color); |
| |
| if (x0 == x1 && y0 == y1) |
| break; |
| |
| e2 = err; |
| |
| if (e2 >-dx) { |
| err -= dy; |
| x0--; |
| } |
| |
| if (e2 < dy) { |
| err += dx; |
| y0 += sy; |
| } |
| } |
| } |
| |
| static void draw_response(AVFilterContext *ctx, AVFrame *out) |
| { |
| AudioFIRContext *s = ctx->priv; |
| 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; |
| |
| memset(out->data[0], 0, s->h * out->linesize[0]); |
| |
| phase = av_malloc_array(s->w, sizeof(*phase)); |
| mag = av_malloc_array(s->w, sizeof(*mag)); |
| delay = av_malloc_array(s->w, sizeof(*delay)); |
| if (!mag || !phase || !delay) |
| goto end; |
| |
| 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->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.; |
| |
| 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, 2, "Max Magnitude:", 0xDDDDDDDD); |
| snprintf(text, sizeof(text), "%.2f", max); |
| drawtext(out, 15 * 8 + 2, 2, text, 0xDDDDDDDD); |
| |
| 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 init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, |
| int offset, int nb_partitions, int part_size) |
| { |
| 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->rdft || !seg->irdft) |
| 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 (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); |
| } |
| |
| 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); |
| |
| 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, cur_nb_taps; |
| float power = 0; |
| |
| 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); |
| |
| 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 + 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 += 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 = 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(cur_nb_taps, 4)); |
| } |
| |
| 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; |
| |
| 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) * 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; |
| } |
| |
| 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); |
| } |
| } |
| |
| s->have_coeffs = 1; |
| |
| return 0; |
| } |
| |
| static int check_ir(AVFilterLink *link) |
| { |
| AVFilterContext *ctx = link->dst; |
| AudioFIRContext *s = ctx->priv; |
| int nb_taps, max_nb_taps; |
| |
| 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 AVERROR(EINVAL); |
| } |
| |
| return 0; |
| } |
| |
| static int activate(AVFilterContext *ctx) |
| { |
| AudioFIRContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| int ret, status, available, wanted; |
| AVFrame *in = NULL; |
| 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[s->selir]) { |
| ret = check_ir(ctx->inputs[1 + s->selir]); |
| if (ret < 0) |
| return ret; |
| |
| if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF) |
| s->eof_coeffs[s->selir] = 1; |
| |
| 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[s->selir]) { |
| ret = convert_coeffs(ctx); |
| 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 (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); |
| } |
| } |
| |
| if (ff_inlink_queued_samples(ctx->inputs[0]) >= s->min_part_size) { |
| ff_filter_set_ready(ctx, 10); |
| return 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; |
| } |
| } |
| |
| if (ff_outlink_frame_wanted(ctx->outputs[0]) && |
| !ff_outlink_get_status(ctx->inputs[0])) { |
| ff_inlink_request_frame(ctx->inputs[0]); |
| return 0; |
| } |
| |
| 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 FFERROR_NOT_READY; |
| } |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| AudioFIRContext *s = ctx->priv; |
| AVFilterFormats *formats; |
| AVFilterChannelLayouts *layouts; |
| static const enum AVSampleFormat sample_fmts[] = { |
| AV_SAMPLE_FMT_FLTP, |
| AV_SAMPLE_FMT_NONE |
| }; |
| static const enum AVPixelFormat pix_fmts[] = { |
| AV_PIX_FMT_RGB0, |
| AV_PIX_FMT_NONE |
| }; |
| int ret; |
| |
| if (s->response) { |
| AVFilterLink *videolink = ctx->outputs[1]; |
| formats = ff_make_format_list(pix_fmts); |
| if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0) |
| return ret; |
| } |
| |
| layouts = ff_all_channel_counts(); |
| 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; |
| |
| 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); |
| if ((ret = ff_set_common_formats(ctx, formats)) < 0) |
| return ret; |
| |
| formats = ff_all_samplerates(); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static int config_output(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFIRContext *s = ctx->priv; |
| |
| 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 + s->selir]->channels; |
| s->pts = AV_NOPTS_VALUE; |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| AudioFIRContext *s = ctx->priv; |
| |
| for (int i = 0; i < s->nb_segments; i++) { |
| uninit_segment(ctx, &s->seg[i]); |
| } |
| |
| av_freep(&s->fdsp); |
| |
| for (int i = 0; i < s->nb_irs; i++) { |
| av_frame_free(&s->ir[i]); |
| } |
| |
| for (unsigned i = 1; i < ctx->nb_inputs; i++) |
| av_freep(&ctx->input_pads[i].name); |
| |
| av_frame_free(&s->video); |
| } |
| |
| static int config_video(AVFilterLink *outlink) |
| { |
| AVFilterContext *ctx = outlink->src; |
| AudioFIRContext *s = ctx->priv; |
| |
| 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); |
| if (!s->video) |
| return AVERROR(ENOMEM); |
| |
| 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 = "main", |
| .type = AVMEDIA_TYPE_AUDIO, |
| }; |
| |
| ret = ff_insert_inpad(ctx, 0, &pad); |
| if (ret < 0) |
| return ret; |
| |
| for (int n = 0; n < s->nb_irs; n++) { |
| pad = (AVFilterPad) { |
| .name = av_asprintf("ir%d", n), |
| .type = AVMEDIA_TYPE_AUDIO, |
| }; |
| |
| 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) |
| 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) |
| return ret; |
| } |
| |
| s->fdsp = avpriv_float_dsp_alloc(0); |
| if (!s->fdsp) |
| return AVERROR(ENOMEM); |
| |
| ff_afir_init(&s->afirdsp); |
| |
| return 0; |
| } |
| |
| 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) |
| |
| static const AVOption afir_options[] = { |
| { "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 }, |
| { "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 = { |
| .name = "afir", |
| .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, |
| .process_command = process_command, |
| .flags = AVFILTER_FLAG_DYNAMIC_INPUTS | |
| AVFILTER_FLAG_DYNAMIC_OUTPUTS | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |
