X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavfilter%2Faf_afir.c;h=1b4e0a0c9e6d099ab7d796921b5ef0c3f16f6779;hb=cb9dbc60db1847fcb594684b92334da54ea8757e;hp=31919f62e9c93d684dbf6b78140d6c71bd672e3e;hpb=ecb5546e97c479378e057f527cc1d79a46c31cb1;p=ffmpeg diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 31919f62e9c..1b4e0a0c9e6 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -25,6 +25,7 @@ #include +#include "libavutil/avstring.h" #include "libavutil/common.h" #include "libavutil/float_dsp.h" #include "libavutil/intreadwrite.h" @@ -56,10 +57,27 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le 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[0]->extended_data[ch] + offset; + 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; @@ -70,8 +88,13 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset) float *dst = (float *)seg->output->extended_data[ch]; float *sum = (float *)seg->sum->extended_data[ch]; - s->fdsp->vector_fmul_scalar(src + seg->input_offset, in, s->dry_gain, FFALIGN(nb_samples, 4)); - emms_c(); + 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) { @@ -80,6 +103,30 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset) 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]; } @@ -114,9 +161,7 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset) av_rdft_calc(seg->irdft[ch], sum); buf = (float *)seg->buffer->extended_data[ch]; - for (n = 0; n < seg->part_size; n++) { - buf[n] += sum[n]; - } + fir_fadd(s, buf, sum, seg->part_size); memcpy(dst, buf, seg->part_size * sizeof(*dst)); @@ -127,13 +172,16 @@ static int fir_quantum(AVFilterContext *ctx, AVFrame *out, int ch, int offset) 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]; - } + fir_fadd(s, ptr, dst, nb_samples); } - s->fdsp->vector_fmul_scalar(ptr, ptr, s->wet_gain, FFALIGN(nb_samples, 4)); - emms_c(); + 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; } @@ -175,7 +223,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) if (s->pts == AV_NOPTS_VALUE) s->pts = in->pts; - s->in[0] = in; + s->in = in; ctx->internal->execute(ctx, fir_channels, out, NULL, FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); @@ -184,7 +232,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); av_frame_free(&in); - s->in[0] = NULL; + s->in = NULL; return ff_filter_frame(outlink, out); } @@ -255,9 +303,9 @@ static void draw_response(AVFilterContext *ctx, AVFrame *out) if (!mag || !phase || !delay) goto end; - channel = av_clip(s->ir_channel, 0, s->in[1]->channels - 1); + channel = av_clip(s->ir_channel, 0, s->ir[s->selir]->channels - 1); for (i = 0; i < s->w; i++) { - const float *src = (const float *)s->in[1]->extended_data[channel]; + const float *src = (const float *)s->ir[s->selir]->extended_data[channel]; double w = i * M_PI / (s->w - 1); double div, real_num = 0., imag_num = 0., real = 0., imag = 0.; @@ -350,7 +398,7 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, if (!seg->part_index || !seg->output_offset) return AVERROR(ENOMEM); - for (int ch = 0; ch < ctx->inputs[0]->channels; ch++) { + for (int ch = 0; ch < ctx->inputs[0]->channels && part_size >= 8; ch++) { seg->rdft[ch] = av_rdft_init(av_log2(2 * part_size), DFT_R2C); seg->irdft[ch] = av_rdft_init(av_log2(2 * part_size), IDFT_C2R); if (!seg->rdft[ch] || !seg->irdft[ch]) @@ -360,7 +408,7 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, 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], seg->nb_partitions * seg->coeff_size * 2); + 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) @@ -369,79 +417,116 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, 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 left, offset = 0, part_size, max_part_size; - int ret, i, ch, n; + int ret, i, ch, n, cur_nb_taps; float power = 0; - s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1]); - if (s->nb_taps <= 0) - return AVERROR(EINVAL); + if (!s->nb_taps) { + int part_size, max_part_size; + int left, offset = 0; - if (s->minp > s->maxp) { - s->maxp = s->minp; - } + s->nb_taps = ff_inlink_queued_samples(ctx->inputs[1 + s->selir]); + if (s->nb_taps <= 0) + return AVERROR(EINVAL); - left = s->nb_taps; - part_size = 1 << av_log2(s->minp); - max_part_size = 1 << av_log2(s->maxp); + 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; - 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); - 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); + } + } - s->nb_segments = i + 1; - ret = init_segment(ctx, &s->seg[i], offset, nb_partitions, 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; - offset += nb_partitions * part_size; - left -= nb_partitions * part_size; - part_size *= 2; - part_size = FFMIN(part_size, max_part_size); + if (ret == 0) + return AVERROR_BUG; } - ret = ff_inlink_consume_samples(ctx->inputs[1], s->nb_taps, s->nb_taps, &s->in[1]); - if (ret < 0) - return ret; - if (ret == 0) - return AVERROR_BUG; - if (s->response) draw_response(ctx, s->video); s->gain = 1; + cur_nb_taps = s->ir[s->selir]->nb_samples; switch (s->gtype) { case -1: /* nothing to do */ break; case 0: - for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) { + float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch]; - for (i = 0; i < s->nb_taps; i++) + for (i = 0; i < cur_nb_taps; i++) power += FFABS(time[i]); } - s->gain = ctx->inputs[1]->channels / power; + s->gain = ctx->inputs[1 + s->selir]->channels / power; break; case 1: - for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) { + float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch]; - for (i = 0; i < s->nb_taps; i++) + for (i = 0; i < cur_nb_taps; i++) power += time[i]; } - s->gain = ctx->inputs[1]->channels / power; + s->gain = ctx->inputs[1 + s->selir]->channels / power; break; case 2: - for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) { + float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch]; - for (i = 0; i < s->nb_taps; i++) + for (i = 0; i < cur_nb_taps; i++) power += time[i] * time[i]; } s->gain = sqrtf(ch / power); @@ -452,17 +537,17 @@ static int convert_coeffs(AVFilterContext *ctx) s->gain = FFMIN(s->gain * s->ir_gain, 1.f); av_log(ctx, AV_LOG_DEBUG, "power %f, gain %f\n", power, s->gain); - for (ch = 0; ch < ctx->inputs[1]->channels; ch++) { - float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) { + float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch]; - s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(s->nb_taps, 4)); + s->fdsp->vector_fmul_scalar(time, time, s->gain, FFALIGN(cur_nb_taps, 4)); } - av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps); + 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]->channels; ch++) { - float *time = (float *)s->in[1]->extended_data[!s->one2many * ch]; + for (ch = 0; ch < ctx->inputs[1 + s->selir]->channels; ch++) { + float *time = (float *)s->ir[s->selir]->extended_data[!s->one2many * ch]; int toffset = 0; for (i = FFMAX(1, s->length * s->nb_taps); i < s->nb_taps; i++) @@ -483,6 +568,14 @@ static int convert_coeffs(AVFilterContext *ctx) 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)); @@ -510,13 +603,12 @@ static int convert_coeffs(AVFilterContext *ctx) } } - av_frame_free(&s->in[1]); s->have_coeffs = 1; return 0; } -static int check_ir(AVFilterLink *link, AVFrame *frame) +static int check_ir(AVFilterLink *link) { AVFilterContext *ctx = link->dst; AudioFIRContext *s = ctx->priv; @@ -543,26 +635,24 @@ static int activate(AVFilterContext *ctx) FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); if (s->response) FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[1], ctx); - if (!s->eof_coeffs) { - AVFrame *ir = NULL; - - ret = check_ir(ctx->inputs[1], ir); + if (!s->eof_coeffs[s->selir]) { + ret = check_ir(ctx->inputs[1 + s->selir]); if (ret < 0) return ret; - if (ff_outlink_get_status(ctx->inputs[1]) == AVERROR_EOF) - s->eof_coeffs = 1; + if (ff_outlink_get_status(ctx->inputs[1 + s->selir]) == AVERROR_EOF) + s->eof_coeffs[s->selir] = 1; - if (!s->eof_coeffs) { + if (!s->eof_coeffs[s->selir]) { if (ff_outlink_frame_wanted(ctx->outputs[0])) - ff_inlink_request_frame(ctx->inputs[1]); + ff_inlink_request_frame(ctx->inputs[1 + s->selir]); else if (s->response && ff_outlink_frame_wanted(ctx->outputs[1])) - ff_inlink_request_frame(ctx->inputs[1]); + ff_inlink_request_frame(ctx->inputs[1 + s->selir]); return 0; } } - if (!s->have_coeffs && s->eof_coeffs) { + if (!s->have_coeffs && s->eof_coeffs[s->selir]) { ret = convert_coeffs(ctx); if (ret < 0) return ret; @@ -582,8 +672,12 @@ static int activate(AVFilterContext *ctx) int64_t new_pts = av_rescale_q(s->pts, ctx->inputs[0]->time_base, ctx->outputs[1]->time_base); if (ff_outlink_frame_wanted(ctx->outputs[1]) && old_pts < new_pts) { + AVFrame *clone; s->video->pts = new_pts; - return ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video)); + clone = av_frame_clone(s->video); + if (!clone) + return AVERROR(ENOMEM); + return ff_filter_frame(ctx->outputs[1], clone); } } @@ -635,7 +729,7 @@ static int query_formats(AVFilterContext *ctx) 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; } @@ -650,16 +744,18 @@ static int query_formats(AVFilterContext *ctx) } else { AVFilterChannelLayouts *mono = NULL; - ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO); - if (ret) - return ret; - - if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts)) < 0) + 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]->in_channel_layouts)) < 0) + if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts)) < 0) return ret; - if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[1]->out_channel_layouts)) < 0) + + ret = ff_add_channel_layout(&mono, AV_CH_LAYOUT_MONO); + if (ret) return ret; + for (int i = 1; i < ctx->nb_inputs; i++) { + if ((ret = ff_channel_layouts_ref(mono, &ctx->inputs[i]->outcfg.channel_layouts)) < 0) + return ret; + } } formats = ff_make_format_list(sample_fmts); @@ -675,49 +771,19 @@ static int config_output(AVFilterLink *outlink) AVFilterContext *ctx = outlink->src; AudioFIRContext *s = ctx->priv; - s->one2many = ctx->inputs[1]->channels == 1; + s->one2many = ctx->inputs[1 + s->selir]->channels == 1; outlink->sample_rate = ctx->inputs[0]->sample_rate; outlink->time_base = ctx->inputs[0]->time_base; outlink->channel_layout = ctx->inputs[0]->channel_layout; outlink->channels = ctx->inputs[0]->channels; s->nb_channels = outlink->channels; - s->nb_coef_channels = ctx->inputs[1]->channels; + s->nb_coef_channels = ctx->inputs[1 + s->selir]->channels; s->pts = AV_NOPTS_VALUE; return 0; } -static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) -{ - 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 av_cold void uninit(AVFilterContext *ctx) { AudioFIRContext *s = ctx->priv; @@ -727,10 +793,14 @@ static av_cold void uninit(AVFilterContext *ctx) } av_freep(&s->fdsp); - av_frame_free(&s->in[1]); - for (int i = 0; i < ctx->nb_outputs; i++) - av_freep(&ctx->output_pads[i].name); + 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); } @@ -767,37 +837,51 @@ static av_cold int init(AVFilterContext *ctx) AVFilterPad pad, vpad; int ret; - pad = (AVFilterPad){ - .name = av_strdup("default"), - .type = AVMEDIA_TYPE_AUDIO, - .config_props = config_output, + pad = (AVFilterPad) { + .name = "main", + .type = AVMEDIA_TYPE_AUDIO, }; - if (!pad.name) - return AVERROR(ENOMEM); + ret = ff_insert_inpad(ctx, 0, &pad); + if (ret < 0) + return ret; - if (s->response) { - vpad = (AVFilterPad){ - .name = av_strdup("filter_response"), - .type = AVMEDIA_TYPE_VIDEO, - .config_props = config_video, + for (int n = 0; n < s->nb_irs; n++) { + pad = (AVFilterPad) { + .name = av_asprintf("ir%d", n), + .type = AVMEDIA_TYPE_AUDIO, }; - if (!vpad.name) + + if (!pad.name) return AVERROR(ENOMEM); + + ret = ff_insert_inpad(ctx, n + 1, &pad); + if (ret < 0) { + av_freep(&pad.name); + return ret; + } } + pad = (AVFilterPad) { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }; + ret = ff_insert_outpad(ctx, 0, &pad); - if (ret < 0) { - av_freep(&pad.name); + if (ret < 0) return ret; - } if (s->response) { + vpad = (AVFilterPad){ + .name = "filter_response", + .type = AVMEDIA_TYPE_VIDEO, + .config_props = config_video, + }; + ret = ff_insert_outpad(ctx, 1, &vpad); - if (ret < 0) { - av_freep(&vpad.name); + if (ret < 0) return ret; - } } s->fdsp = avpriv_float_dsp_alloc(0); @@ -809,18 +893,31 @@ static av_cold int init(AVFilterContext *ctx) return 0; } -static const AVFilterPad afir_inputs[] = { - { - .name = "main", - .type = AVMEDIA_TYPE_AUDIO, - },{ - .name = "ir", - .type = AVMEDIA_TYPE_AUDIO, - }, - { NULL } -}; +static int process_command(AVFilterContext *ctx, + const char *cmd, + const char *arg, + char *res, + int res_len, + int flags) +{ + AudioFIRContext *s = ctx->priv; + int prev_ir = s->selir; + int ret = ff_filter_process_command(ctx, cmd, arg, res, res_len, flags); + + if (ret < 0) + return ret; + + s->selir = FFMIN(s->nb_irs - 1, s->selir); + + if (prev_ir != s->selir) { + s->have_coeffs = 0; + } + + return 0; +} #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM +#define AFR AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM #define VF AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM #define OFFSET(x) offsetof(AudioFIRContext, x) @@ -842,8 +939,10 @@ static const AVOption afir_options[] = { { "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}, 8, 32768, AF }, + { "minp", "set min partition size", OFFSET(minp), AV_OPT_TYPE_INT, {.i64=8192}, 1, 32768, AF }, { "maxp", "set max partition size", OFFSET(maxp), AV_OPT_TYPE_INT, {.i64=8192}, 8, 32768, AF }, + { "nbirs", "set number of input IRs",OFFSET(nb_irs),AV_OPT_TYPE_INT, {.i64=1}, 1, 32, AF }, + { "ir", "select IR", OFFSET(selir), AV_OPT_TYPE_INT, {.i64=0}, 0, 31, AFR }, { NULL } }; @@ -851,14 +950,15 @@ AVFILTER_DEFINE_CLASS(afir); 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, };