/* * 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 #include "libavutil/cpu.h" #include "libavutil/tx.h" #include "libavutil/avstring.h" #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/frame.h" #include "libavutil/intreadwrite.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "libavutil/rational.h" #include "libavutil/xga_font_data.h" #include "audio.h" #include "avfilter.h" #include "filters.h" #include "formats.h" #include "internal.h" #include "af_afir.h" #include "af_afirdsp.h" 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; } } } #define DEPTH 32 #include "afir_template.c" #undef DEPTH #define DEPTH 64 #include "afir_template.c" static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch) { AudioFIRContext *s = ctx->priv; const int min_part_size = s->min_part_size; const int prev_selir = s->prev_selir; const int selir = s->selir; for (int offset = 0; offset < out->nb_samples; offset += min_part_size) { switch (s->format) { case AV_SAMPLE_FMT_FLTP: if (prev_selir != selir) { const float *xfade0 = (const float *)s->xfade[0]->extended_data[ch]; const float *xfade1 = (const float *)s->xfade[1]->extended_data[ch]; float *src0 = (float *)s->fadein[0]->extended_data[ch]; float *src1 = (float *)s->fadein[1]->extended_data[ch]; float *dst = ((float *)out->extended_data[ch]) + offset; memset(src0, 0, min_part_size * sizeof(float)); memset(src1, 0, min_part_size * sizeof(float)); fir_quantum_float(ctx, s->fadein[0], ch, offset, 0, prev_selir); fir_quantum_float(ctx, s->fadein[1], ch, offset, 0, selir); if (s->loading[ch] > 0 && s->loading[ch] <= min_part_size) { for (int n = 0; n < min_part_size; n++) dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n]; s->loading[ch] = 0; } else { memcpy(dst, src0, min_part_size * sizeof(float)); } } else { fir_quantum_float(ctx, out, ch, offset, offset, selir); } break; case AV_SAMPLE_FMT_DBLP: if (prev_selir != selir) { const double *xfade0 = (const double *)s->xfade[0]->extended_data[ch]; const double *xfade1 = (const double *)s->xfade[1]->extended_data[ch]; double *src0 = (double *)s->fadein[0]->extended_data[ch]; double *src1 = (double *)s->fadein[1]->extended_data[ch]; double *dst = ((double *)out->extended_data[ch]) + offset; memset(src0, 0, min_part_size * sizeof(double)); memset(src1, 0, min_part_size * sizeof(double)); fir_quantum_double(ctx, s->fadein[0], ch, offset, 0, prev_selir); fir_quantum_double(ctx, s->fadein[1], ch, offset, 0, selir); if (s->loading[ch] > 0 && s->loading[ch] <= min_part_size) { for (int n = 0; n < min_part_size; n++) dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n]; s->loading[ch] = 0; } else { memcpy(dst, src0, min_part_size * sizeof(double)); } } else { fir_quantum_double(ctx, out, ch, offset, offset, selir); } break; } if (selir != prev_selir && s->loading[ch] > 0) s->loading[ch] -= min_part_size; } return 0; } static int fir_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AVFrame *out = arg; const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; const int end = (out->ch_layout.nb_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; out = ff_get_audio_buffer(outlink, in->nb_samples); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); out->pts = s->pts = in->pts; s->in = in; ff_filter_execute(ctx, fir_channels, out, NULL, FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); av_frame_free(&in); s->in = NULL; return ff_filter_frame(outlink, out); } static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir, int offset, int nb_partitions, int part_size, int index) { AudioFIRContext *s = ctx->priv; const size_t cpu_align = av_cpu_max_align(); union { double d; float f; } cscale, scale, iscale; enum AVTXType tx_type; int ret; seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx)); seg->ctx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->ctx)); seg->itx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->itx)); if (!seg->tx || !seg->ctx || !seg->itx) return AVERROR(ENOMEM); seg->fft_length = (part_size + 1) * 2; seg->part_size = part_size; seg->block_size = FFALIGN(seg->fft_length, cpu_align); seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align); 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]->ch_layout.nb_channels, sizeof(*seg->part_index)); seg->output_offset = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->output_offset)); if (!seg->part_index || !seg->output_offset) return AVERROR(ENOMEM); switch (s->format) { case AV_SAMPLE_FMT_FLTP: cscale.f = 1.f; scale.f = 1.f / sqrtf(2.f * part_size); iscale.f = 1.f / sqrtf(2.f * part_size); tx_type = AV_TX_FLOAT_RDFT; break; case AV_SAMPLE_FMT_DBLP: cscale.d = 1.0; scale.d = 1.0 / sqrt(2.0 * part_size); iscale.d = 1.0 / sqrt(2.0 * part_size); tx_type = AV_TX_DOUBLE_RDFT; break; } for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) { ret = av_tx_init(&seg->ctx[ch], &seg->ctx_fn, tx_type, 0, 2 * part_size, &cscale, 0); if (ret < 0) return ret; ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type, 0, 2 * part_size, &scale, 0); if (ret < 0) return ret; ret = av_tx_init(&seg->itx[ch], &seg->itx_fn, tx_type, 1, 2 * part_size, &iscale, 0); if (ret < 0) return ret; } seg->sumin = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length); seg->sumout = ff_get_audio_buffer(ctx->inputs[0], seg->fft_length); seg->blockout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size * seg->nb_partitions); seg->tempin = ff_get_audio_buffer(ctx->inputs[0], seg->block_size); seg->tempout = ff_get_audio_buffer(ctx->inputs[0], seg->block_size); seg->buffer = ff_get_audio_buffer(ctx->inputs[0], seg->part_size); seg->input = ff_get_audio_buffer(ctx->inputs[0], seg->input_size); seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5); if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout || !seg->input || !seg->output || !seg->tempin || !seg->tempout) return AVERROR(ENOMEM); return 0; } static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) { AudioFIRContext *s = ctx->priv; if (seg->ctx) { for (int ch = 0; ch < s->nb_channels; ch++) av_tx_uninit(&seg->ctx[ch]); } av_freep(&seg->ctx); if (seg->tx) { for (int ch = 0; ch < s->nb_channels; ch++) av_tx_uninit(&seg->tx[ch]); } av_freep(&seg->tx); if (seg->itx) { for (int ch = 0; ch < s->nb_channels; ch++) av_tx_uninit(&seg->itx[ch]); } av_freep(&seg->itx); av_freep(&seg->output_offset); av_freep(&seg->part_index); av_frame_free(&seg->tempin); av_frame_free(&seg->tempout); av_frame_free(&seg->blockout); av_frame_free(&seg->sumin); av_frame_free(&seg->sumout); av_frame_free(&seg->buffer); av_frame_free(&seg->input); av_frame_free(&seg->output); seg->input_size = 0; for (int i = 0; i < MAX_IR_STREAMS; i++) av_frame_free(&seg->coeff); } static int convert_coeffs(AVFilterContext *ctx, int selir) { AudioFIRContext *s = ctx->priv; int ret, nb_taps, cur_nb_taps; if (!s->nb_taps[selir]) { int part_size, max_part_size; int left, offset = 0; s->nb_taps[selir] = ff_inlink_queued_samples(ctx->inputs[1 + selir]); if (s->nb_taps[selir] <= 0) return AVERROR(EINVAL); if (s->minp > s->maxp) s->maxp = s->minp; if (s->nb_segments[selir]) goto skip; left = s->nb_taps[selir]; part_size = 1 << av_log2(s->minp); max_part_size = 1 << av_log2(s->maxp); for (int 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[selir] = i + 1; s->max_offset[selir] = offset; ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i); 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); } } skip: if (!s->ir[selir]) { ret = ff_inlink_consume_samples(ctx->inputs[1 + selir], s->nb_taps[selir], s->nb_taps[selir], &s->ir[selir]); if (ret < 0) return ret; if (ret == 0) return AVERROR_BUG; } if (s->response) { switch (s->format) { case AV_SAMPLE_FMT_FLTP: draw_response_float(ctx, s->video); break; case AV_SAMPLE_FMT_DBLP: draw_response_double(ctx, s->video); break; } } cur_nb_taps = s->ir[selir]->nb_samples; nb_taps = cur_nb_taps; if (!s->norm_ir[selir] || s->norm_ir[selir]->nb_samples < nb_taps) { av_frame_free(&s->norm_ir[selir]); s->norm_ir[selir] = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8)); if (!s->norm_ir[selir]) return AVERROR(ENOMEM); } 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[selir]); switch (s->format) { case AV_SAMPLE_FMT_FLTP: for (int ch = 0; ch < s->nb_channels; ch++) { const float *tsrc = (const float *)s->ir[selir]->extended_data[!s->one2many * ch]; float *time = (float *)s->norm_ir[selir]->extended_data[ch]; memcpy(time, tsrc, sizeof(*time) * nb_taps); for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++) time[i] = 0; get_power_float(ctx, s, nb_taps, ch, time); for (int n = 0; n < s->nb_segments[selir]; n++) { AudioFIRSegment *seg = &s->seg[selir][n]; if (!seg->coeff) seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2); if (!seg->coeff) return AVERROR(ENOMEM); for (int i = 0; i < seg->nb_partitions; i++) convert_channel_float(ctx, s, ch, seg, i, selir); } } break; case AV_SAMPLE_FMT_DBLP: for (int ch = 0; ch < s->nb_channels; ch++) { const double *tsrc = (const double *)s->ir[selir]->extended_data[!s->one2many * ch]; double *time = (double *)s->norm_ir[selir]->extended_data[ch]; memcpy(time, tsrc, sizeof(*time) * nb_taps); for (int i = FFMAX(1, s->length * nb_taps); i < nb_taps; i++) time[i] = 0; get_power_double(ctx, s, nb_taps, ch, time); for (int n = 0; n < s->nb_segments[selir]; n++) { AudioFIRSegment *seg = &s->seg[selir][n]; if (!seg->coeff) seg->coeff = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2); if (!seg->coeff) return AVERROR(ENOMEM); for (int i = 0; i < seg->nb_partitions; i++) convert_channel_double(ctx, s, ch, seg, i, selir); } } break; } s->have_coeffs[selir] = 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->selir] && s->eof_coeffs[s->selir]) { ret = convert_coeffs(ctx, s->selir); if (ret < 0) return ret; } if (s->selir != s->prev_selir && s->loading[0] <= 0) { for (int ch = 0; ch < s->nb_channels; ch++) s->loading[ch] = s->max_offset[s->selir] + s->max_part_size; } 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->selir != s->prev_selir && s->loading[0] <= 0) s->prev_selir = s->selir; if (ret < 0) return ret; if (s->response && s->have_coeffs[s->selir]) { 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; static const enum AVSampleFormat sample_fmts[3][3] = { { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, { AV_SAMPLE_FMT_DBLP, 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]; AVFilterFormats *formats = ff_make_format_list(pix_fmts); if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0) return ret; } if (s->ir_format) { ret = ff_set_common_all_channel_counts(ctx); if (ret < 0) return ret; } else { AVFilterChannelLayouts *mono = NULL; AVFilterChannelLayouts *layouts = ff_all_channel_counts(); 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, &(AVChannelLayout)AV_CHANNEL_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; } } if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0) return ret; return ff_set_common_all_samplerates(ctx); } static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; AudioFIRContext *s = ctx->priv; int ret; s->one2many = ctx->inputs[1 + s->selir]->ch_layout.nb_channels == 1; outlink->sample_rate = ctx->inputs[0]->sample_rate; outlink->time_base = ctx->inputs[0]->time_base; #if FF_API_OLD_CHANNEL_LAYOUT FF_DISABLE_DEPRECATION_WARNINGS outlink->channel_layout = ctx->inputs[0]->channel_layout; FF_ENABLE_DEPRECATION_WARNINGS #endif if ((ret = av_channel_layout_copy(&outlink->ch_layout, &ctx->inputs[0]->ch_layout)) < 0) return ret; outlink->ch_layout.nb_channels = ctx->inputs[0]->ch_layout.nb_channels; s->format = outlink->format; s->nb_channels = outlink->ch_layout.nb_channels; s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading)); if (!s->loading) return AVERROR(ENOMEM); s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size); s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size); if (!s->fadein[0] || !s->fadein[1]) return AVERROR(ENOMEM); s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size); s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size); if (!s->xfade[0] || !s->xfade[1]) return AVERROR(ENOMEM); switch (s->format) { case AV_SAMPLE_FMT_FLTP: for (int ch = 0; ch < s->nb_channels; ch++) { float *dst0 = (float *)s->xfade[0]->extended_data[ch]; float *dst1 = (float *)s->xfade[1]->extended_data[ch]; for (int n = 0; n < s->min_part_size; n++) { dst0[n] = (n + 1.f) / s->min_part_size; dst1[n] = 1.f - dst0[n]; } } break; case AV_SAMPLE_FMT_DBLP: for (int ch = 0; ch < s->nb_channels; ch++) { double *dst0 = (double *)s->xfade[0]->extended_data[ch]; double *dst1 = (double *)s->xfade[1]->extended_data[ch]; for (int n = 0; n < s->min_part_size; n++) { dst0[n] = (n + 1.0) / s->min_part_size; dst1[n] = 1.0 - dst0[n]; } } break; } return 0; } static av_cold void uninit(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; av_freep(&s->fdsp); av_freep(&s->loading); for (int i = 0; i < s->nb_irs; i++) { for (int j = 0; j < s->nb_segments[i]; j++) uninit_segment(ctx, &s->seg[i][j]); av_frame_free(&s->ir[i]); av_frame_free(&s->norm_ir[i]); } av_frame_free(&s->fadein[0]); av_frame_free(&s->fadein[1]); av_frame_free(&s->xfade[0]); av_frame_free(&s->xfade[1]); 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; } static av_cold int init(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; AVFilterPad pad, vpad; int ret; s->prev_selir = FFMIN(s->nb_irs - 1, s->selir); pad = (AVFilterPad) { .name = "main", .type = AVMEDIA_TYPE_AUDIO, }; ret = ff_append_inpad(ctx, &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_append_inpad_free_name(ctx, &pad); if (ret < 0) return ret; } pad = (AVFilterPad) { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }; ret = ff_append_outpad(ctx, &pad); if (ret < 0) return ret; if (s->response) { vpad = (AVFilterPad){ .name = "filter_response", .type = AVMEDIA_TYPE_VIDEO, .config_props = config_video, }; ret = ff_append_outpad(ctx, &vpad); if (ret < 0) return ret; } s->fdsp = avpriv_float_dsp_alloc(0); if (!s->fdsp) return AVERROR(ENOMEM); ff_afir_init(&s->afirdsp); s->min_part_size = 1 << av_log2(s->minp); s->max_part_size = 1 << av_log2(s->maxp); 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_selir, ret; prev_selir = s->selir; 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 (s->selir != prev_selir) s->prev_selir = prev_selir; 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, AFR }, { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, AFR }, { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF }, { "gtype", "set IR auto gain type",OFFSET(gtype), AV_OPT_TYPE_INT, {.i64=0}, -1, 4, AF, "gtype" }, { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "gtype" }, { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "gtype" }, { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "gtype" }, { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "gtype" }, { "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "gtype" }, { "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "gtype" }, { "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, 65536, AF }, { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 65536, 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 }, { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" }, { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" }, { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" }, { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" }, { NULL } }; AVFILTER_DEFINE_CLASS(afir); const 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, FILTER_QUERY_FUNC(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, };