| /* |
| * 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/cpu.h" |
| #include "libavutil/mem.h" |
| #include "libavutil/tx.h" |
| #include "libavutil/avstring.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/frame.h" |
| #include "libavutil/log.h" |
| #include "libavutil/opt.h" |
| |
| #include "audio.h" |
| #include "avfilter.h" |
| #include "filters.h" |
| #include "formats.h" |
| #include "internal.h" |
| #include "af_afir.h" |
| #include "af_afirdsp.h" |
| |
| #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: |
| fir_quantums_float(ctx, s, out, min_part_size, ch, offset, prev_selir, selir); |
| break; |
| case AV_SAMPLE_FMT_DBLP: |
| fir_quantums_double(ctx, s, out, min_part_size, ch, offset, prev_selir, 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))); |
| s->prev_is_disabled = ctx->is_disabled; |
| |
| 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->coeff_size = FFALIGN(seg->part_size + 1, cpu_align); |
| seg->block_size = FFMAX(seg->coeff_size * 2, FFALIGN(seg->fft_length, 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; |
| 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; |
| s->max_offset[selir] = offset; |
| 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; |
| } |
| |
| 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]; |
| |
| s->ch_gain[ch] = ir_gain_float(ctx, s, nb_taps, tsrc); |
| } |
| |
| if (s->ir_link) { |
| float gain = +INFINITY; |
| |
| for (int ch = 0; ch < s->nb_channels; ch++) |
| gain = fminf(gain, s->ch_gain[ch]); |
| |
| for (int ch = 0; ch < s->nb_channels; ch++) |
| s->ch_gain[ch] = gain; |
| } |
| |
| 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; |
| |
| ir_scale_float(ctx, s, nb_taps, ch, time, s->ch_gain[ch]); |
| |
| 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]; |
| |
| s->ch_gain[ch] = ir_gain_double(ctx, s, nb_taps, tsrc); |
| } |
| |
| if (s->ir_link) { |
| double gain = +INFINITY; |
| |
| for (int ch = 0; ch < s->nb_channels; ch++) |
| gain = fmin(gain, s->ch_gain[ch]); |
| |
| for (int ch = 0; ch < s->nb_channels; ch++) |
| s->ch_gain[ch] = gain; |
| } |
| |
| 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; |
| |
| ir_scale_double(ctx, s, nb_taps, ch, time, s->ch_gain[ch]); |
| |
| 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, int selir) |
| { |
| 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); |
| } |
| |
| if (ff_inlink_check_available_samples(link, nb_taps + 1) == 1) |
| s->eof_coeffs[selir] = 1; |
| |
| 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); |
| |
| for (int i = 0; i < s->nb_irs; i++) { |
| const int selir = i; |
| |
| if (s->ir_load && selir != s->selir) |
| continue; |
| |
| if (!s->eof_coeffs[selir]) { |
| ret = check_ir(ctx->inputs[1 + selir], selir); |
| if (ret < 0) |
| return ret; |
| |
| if (!s->eof_coeffs[selir]) { |
| if (ff_outlink_frame_wanted(ctx->outputs[0])) |
| ff_inlink_request_frame(ctx->inputs[1 + selir]); |
| return 0; |
| } |
| } |
| |
| if (!s->have_coeffs[selir] && s->eof_coeffs[selir]) { |
| ret = convert_coeffs(ctx, selir); |
| 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->selir != s->prev_selir && s->loading[0] == 0) |
| s->prev_selir = s->selir; |
| |
| if (ret < 0) |
| return ret; |
| |
| 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); |
| return 0; |
| } |
| } |
| |
| if (ff_outlink_frame_wanted(ctx->outputs[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 }, |
| }; |
| int 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 ((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->ch_gain = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->ch_gain)); |
| s->loading = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*s->loading)); |
| if (!s->loading || !s->ch_gain) |
| 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->ch_gain); |
| 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]); |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| AudioFIRContext *s = ctx->priv; |
| AVFilterPad pad; |
| 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; |
| } |
| |
| 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; |
| |
| for (int ch = 0; ch < s->nb_channels; ch++) |
| s->loading[ch] = 1; |
| } |
| |
| 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|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "none", "without auto gain", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "peak", "peak gain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "dc", "DC gain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "gn", "gain to noise", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "ac", "AC gain", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "rms", "RMS gain", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF|AV_OPT_FLAG_DEPRECATED, .unit = "gtype" }, |
| { "irnorm", "set IR norm", OFFSET(ir_norm), AV_OPT_TYPE_FLOAT, {.dbl=1}, -1, 2, AF }, |
| { "irlink", "set IR link", OFFSET(ir_link), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF }, |
| { "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, .unit = "irfmt" }, |
| { "mono", "single channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irfmt" }, |
| { "input", "same as input", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "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|AV_OPT_FLAG_DEPRECATED }, |
| { "channel", "set IR channel to display frequency response", OFFSET(ir_channel), AV_OPT_TYPE_INT, {.i64=0}, 0, 1024, VF|AV_OPT_FLAG_DEPRECATED }, |
| { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, VF|AV_OPT_FLAG_DEPRECATED }, |
| { "rate", "set video rate", OFFSET(frame_rate), AV_OPT_TYPE_VIDEO_RATE, {.str = "25"}, 0, INT32_MAX, VF|AV_OPT_FLAG_DEPRECATED }, |
| { "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, .unit = "precision" }, |
| { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" }, |
| { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" }, |
| { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" }, |
| { "irload", "set IR loading type", OFFSET(ir_load), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, .unit = "irload" }, |
| { "init", "load all IRs on init", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "irload" }, |
| { "access", "load IR on access", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "irload" }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(afir); |
| |
| static const AVFilterPad outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_output, |
| }, |
| }; |
| |
| 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), |
| FILTER_OUTPUTS(outputs), |
| .init = init, |
| .activate = activate, |
| .uninit = uninit, |
| .process_command = process_command, |
| .flags = AVFILTER_FLAG_DYNAMIC_INPUTS | |
| AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | |
| AVFILTER_FLAG_SLICE_THREADS, |
| }; |