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author | Paul B Mahol <onemda@gmail.com> | 2022-12-31 23:31:31 +0100 |
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committer | Paul B Mahol <onemda@gmail.com> | 2023-01-02 15:33:57 +0100 |
commit | 3879555cd540f7df02ef527fcbc0fda4c68fbfa9 (patch) | |
tree | 3a45b52c2a17815762ceb34d1a389ee1fa6c2e36 /libavfilter/af_afir.c | |
parent | 62da0b4a741a064f118a0eece496d6bcc437ec91 (diff) | |
download | ffmpeg-3879555cd540f7df02ef527fcbc0fda4c68fbfa9.tar.gz |
avfilter/afir_template: make IR transitions artifacts free
Diffstat (limited to 'libavfilter/af_afir.c')
-rw-r--r-- | libavfilter/af_afir.c | 84 |
1 files changed, 66 insertions, 18 deletions
diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c index 2d09b2a0e1..11fa5074d0 100644 --- a/libavfilter/af_afir.c +++ b/libavfilter/af_afir.c @@ -155,7 +155,7 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink) return ff_filter_frame(outlink, out); } -static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, +static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, int selir, int offset, int nb_partitions, int part_size, int index) { AudioFIRContext *s = ctx->priv; @@ -221,12 +221,10 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg, 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->coeff = ff_get_audio_buffer(ctx->inputs[0], 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); - seg->loaded = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions); + seg->output = ff_get_audio_buffer(ctx->inputs[0], seg->part_size * 5); if (!seg->buffer || !seg->sumin || !seg->sumout || !seg->blockout || - !seg->coeff || !seg->input || !seg->output || !seg->loaded || !seg->tempin || !seg->tempout) + !seg->input || !seg->output || !seg->tempin || !seg->tempout) return AVERROR(ENOMEM); return 0; @@ -264,18 +262,18 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg) av_frame_free(&seg->sumin); av_frame_free(&seg->sumout); av_frame_free(&seg->buffer); - av_frame_free(&seg->coeff); av_frame_free(&seg->input); av_frame_free(&seg->output); - av_frame_free(&seg->loaded); seg->input_size = 0; + + for (int i = 0; i < MAX_IR_STREAMS; i++) + av_frame_free(&seg->coeff[i]); } static int convert_coeffs(AVFilterContext *ctx, int selir) { AudioFIRContext *s = ctx->priv; - const int prev_selir = s->prev_selir; - int ret, nb_taps, cur_nb_taps, prev_nb_taps; + int ret, nb_taps, cur_nb_taps; if (!s->nb_taps[selir]) { int part_size, max_part_size; @@ -302,7 +300,7 @@ static int convert_coeffs(AVFilterContext *ctx, int selir) 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, i); + ret = init_segment(ctx, &s->seg[i], selir, offset, nb_partitions, part_size, i); if (ret < 0) return ret; offset += nb_partitions * part_size; @@ -333,19 +331,68 @@ skip: } cur_nb_taps = s->ir[selir]->nb_samples; - prev_nb_taps = s->ir[prev_selir]->nb_samples; - nb_taps = FFMAX(cur_nb_taps, prev_nb_taps); + nb_taps = cur_nb_taps; - if (!s->norm_ir || s->norm_ir->nb_samples < nb_taps) { - av_frame_free(&s->norm_ir); - s->norm_ir = ff_get_audio_buffer(ctx->inputs[0], FFALIGN(nb_taps, 8)); - if (!s->norm_ir) + 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); + 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; n++) { + AudioFIRSegment *seg = &s->seg[n]; + + if (!seg->coeff[selir]) + seg->coeff[selir] = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2); + if (!seg->coeff[selir]) + 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; n++) { + AudioFIRSegment *seg = &s->seg[n]; + + if (!seg->coeff[selir]) + seg->coeff[selir] = ff_get_audio_buffer(ctx->inputs[0], seg->nb_partitions * seg->coeff_size * 2); + if (!seg->coeff[selir]) + 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; @@ -536,9 +583,10 @@ static av_cold void uninit(AVFilterContext *ctx) av_freep(&s->fdsp); - av_frame_free(&s->norm_ir); - for (int i = 0; i < s->nb_irs; i++) + for (int i = 0; i < s->nb_irs; i++) { av_frame_free(&s->ir[i]); + av_frame_free(&s->norm_ir[i]); + } av_frame_free(&s->video); } |