diff --git a/doc/filters.texi b/doc/filters.texi index dc8bb54994..468c277798 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -3500,6 +3500,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Commands @@ -3589,6 +3596,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Commands @@ -3688,6 +3702,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Commands @@ -3772,6 +3793,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @section bs2b @@ -4566,6 +4594,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Examples @@ -5069,6 +5104,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Commands @@ -5421,6 +5463,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Examples @@ -6659,6 +6708,13 @@ Always use float 32-bit. @item f64 Always use float 64-bit. @end table + +@item block_size, b +Set block size used for reverse IIR processing. If this value is set to high enough +value (higher than impulse response length truncated when reaches near zero values) filtering +will become linear phase otherwise if not big enough it will just produce nasty artifacts. + +Note that filter delay will be exactly this many samples when set to non-zero value. @end table @subsection Commands diff --git a/libavfilter/af_biquads.c b/libavfilter/af_biquads.c index 8caf169d50..2ec32e915d 100644 --- a/libavfilter/af_biquads.c +++ b/libavfilter/af_biquads.c @@ -70,6 +70,7 @@ #include "libavutil/opt.h" #include "audio.h" #include "avfilter.h" +#include "filters.h" #include "internal.h" enum FilterType { @@ -109,6 +110,8 @@ enum TransformType { typedef struct ChanCache { double i1, i2; double o1, o2; + double ri1, ri2; + double ro1, ro2; int clippings; } ChanCache; @@ -121,6 +124,7 @@ typedef struct BiquadsContext { int csg; int transform_type; int precision; + int block_samples; int bypass; @@ -139,6 +143,8 @@ typedef struct BiquadsContext { double oa0, oa1, oa2; double ob0, ob1, ob2; + AVFrame *block[3]; + ChanCache *cache; int block_align; @@ -782,6 +788,14 @@ static int config_filter(AVFilterLink *outlink, int reset) if (reset) memset(s->cache, 0, sizeof(ChanCache) * inlink->ch_layout.nb_channels); + if (reset && s->block_samples > 0) { + for (int i = 0; i < 3; i++) { + s->block[i] = ff_get_audio_buffer(outlink, s->block_samples * 2); + if (!s->block[i]) + return AVERROR(ENOMEM); + } + } + switch (s->transform_type) { case DI: switch (inlink->format) { @@ -908,6 +922,41 @@ typedef struct ThreadData { AVFrame *in, *out; } ThreadData; +static void reverse_samples(AVFrame *out, AVFrame *in, int p, + int oo, int io, int nb_samples) +{ + switch (out->format) { + case AV_SAMPLE_FMT_S16P: { + const int16_t *src = ((const int16_t *)out->extended_data[p]) + io; + int16_t *dst = ((int16_t *)out->extended_data[p]) + oo; + for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--) + dst[i] = src[j]; + } + break; + case AV_SAMPLE_FMT_S32P: { + const int32_t *src = ((const int32_t *)out->extended_data[p]) + io; + int32_t *dst = ((int32_t *)out->extended_data[p]) + oo; + for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--) + dst[i] = src[j]; + } + break; + case AV_SAMPLE_FMT_FLTP: { + const float *src = ((const float *)in->extended_data[p]) + io; + float *dst = ((float *)out->extended_data[p]) + oo; + for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--) + dst[i] = src[j]; + } + break; + case AV_SAMPLE_FMT_DBLP: { + const double *src = ((const double *)in->extended_data[p]) + io; + double *dst = ((double *)out->extended_data[p]) + oo; + for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--) + dst[i] = src[j]; + } + break; + } +} + static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AVFilterLink *inlink = ctx->inputs[0]; @@ -930,9 +979,37 @@ static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_job continue; } - s->filter(s, buf->extended_data[ch], out_buf->extended_data[ch], buf->nb_samples, - &s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2, - s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled); + if (!s->block_samples) { + s->filter(s, buf->extended_data[ch], out_buf->extended_data[ch], buf->nb_samples, + &s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2, + s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled); + } else { + memcpy(s->block[0]->extended_data[ch] + s->block_align * s->block_samples, buf->extended_data[ch], + buf->nb_samples * s->block_align); + s->filter(s, s->block[0]->extended_data[ch], s->block[1]->extended_data[ch], s->block_samples, + &s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2, + s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled); + s->cache[ch].ri1 = s->cache[ch].i1; + s->cache[ch].ri2 = s->cache[ch].i2; + s->cache[ch].ro1 = s->cache[ch].o1; + s->cache[ch].ro2 = s->cache[ch].o2; + s->filter(s, s->block[0]->extended_data[ch] + s->block_samples * s->block_align, + s->block[1]->extended_data[ch] + s->block_samples * s->block_align, + s->block_samples, + &s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2, + s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled); + reverse_samples(s->block[2], s->block[1], ch, 0, 0, s->block_samples * 2); + s->cache[ch].ri1 = 0.; + s->cache[ch].ri2 = 0.; + s->cache[ch].ro1 = 0.; + s->cache[ch].ro2 = 0.; + s->filter(s, s->block[2]->extended_data[ch], s->block[2]->extended_data[ch], s->block[2]->nb_samples, + &s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2, + s->b0, s->b1, s->b2, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled); + reverse_samples(out_buf, s->block[2], ch, 0, s->block_samples, out_buf->nb_samples); + memmove(s->block[0]->extended_data[ch], s->block[0]->extended_data[ch] + s->block_align * s->block_samples, + s->block_samples * s->block_align); + } } return 0; @@ -988,6 +1065,37 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf) return ff_filter_frame(outlink, out_buf); } +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + BiquadsContext *s = ctx->priv; + AVFrame *in = NULL; + int ret; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + if (s->block_samples > 0) { + ret = ff_inlink_consume_samples(inlink, s->block_samples, s->block_samples, &in); + } else { + ret = ff_inlink_consume_frame(inlink, &in); + } + if (ret < 0) + return ret; + if (ret > 0) + return filter_frame(inlink, in); + + if (s->block_samples > 0 && ff_inlink_queued_samples(inlink) >= s->block_samples) { + ff_filter_set_ready(ctx, 10); + return 0; + } + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + FF_FILTER_FORWARD_WANTED(outlink, inlink); + + return FFERROR_NOT_READY; +} + static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags) { @@ -1005,6 +1113,8 @@ static av_cold void uninit(AVFilterContext *ctx) { BiquadsContext *s = ctx->priv; + for (int i = 0; i < 3; i++) + av_frame_free(&s->block[i]); av_freep(&s->cache); av_channel_layout_uninit(&s->ch_layout); } @@ -1013,7 +1123,6 @@ static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, }, }; @@ -1043,6 +1152,7 @@ const AVFilter ff_af_##name_ = { \ .priv_class = &priv_class_##_class, \ .priv_size = sizeof(BiquadsContext), \ .init = name_##_init, \ + .activate = activate, \ .uninit = uninit, \ FILTER_INPUTS(inputs), \ FILTER_OUTPUTS(outputs), \ @@ -1091,6 +1201,8 @@ static const AVOption equalizer_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1134,6 +1246,8 @@ static const AVOption bass_lowshelf_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1184,6 +1298,8 @@ static const AVOption treble_highshelf_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1233,6 +1349,8 @@ static const AVOption bandpass_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1272,6 +1390,8 @@ static const AVOption bandreject_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1313,6 +1433,8 @@ static const AVOption lowpass_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1354,6 +1476,8 @@ static const AVOption highpass_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} }; @@ -1429,6 +1553,8 @@ static const AVOption biquad_options[] = { {"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, {"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, {"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}, + {"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, + {"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF}, {NULL} };