forked from FFmpeg/FFmpeg
avfilter/af_earwax: fix filter behavior
Previous filter output was incorrect. New one actually follows graph in comments described on side of filter taps.
This commit is contained in:
Paul B Mahol
parent
c369ceb244
commit
f41de0436c
3 changed files with 118 additions and 50 deletions
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@ -34,9 +34,9 @@
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#include "audio.h"
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#include "formats.h"
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#define NUMTAPS 64
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#define NUMTAPS 32
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static const int8_t filt[NUMTAPS] = {
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static const int8_t filt[NUMTAPS * 2] = {
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/* 30° 330° */
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4, -6, /* 32 tap stereo FIR filter. */
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4, -11, /* One side filters as if the */
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@ -72,7 +72,10 @@ static const int8_t filt[NUMTAPS] = {
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4, 0};
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typedef struct EarwaxContext {
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int16_t taps[NUMTAPS * 2];
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int16_t filter[2][NUMTAPS];
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int16_t taps[4][NUMTAPS * 2];
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AVFrame *frame[2];
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} EarwaxContext;
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static int query_formats(AVFilterContext *ctx)
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@ -83,7 +86,7 @@ static int query_formats(AVFilterContext *ctx)
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 ||
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
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@ -94,7 +97,8 @@ static int query_formats(AVFilterContext *ctx)
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}
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//FIXME: replace with DSPContext.scalarproduct_int16
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static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
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static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
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const int16_t *filt, int16_t *out)
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{
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int32_t sample;
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int16_t j;
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@ -103,7 +107,7 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
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sample = 0;
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for (j = 0; j < NUMTAPS; j++)
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sample += in[j] * filt[j];
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*out = av_clip_int16(sample >> 6);
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*out = av_clip_int16(sample >> 7);
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out++;
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in++;
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}
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@ -111,40 +115,102 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
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return out;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterLink *outlink = inlink->dst->outputs[0];
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int16_t *taps, *endin, *in, *out;
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AVFrame *outsamples = ff_get_audio_buffer(outlink, insamples->nb_samples);
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EarwaxContext *s = inlink->dst->priv;
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for (int i = 0; i < NUMTAPS; i++) {
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s->filter[0][i] = filt[i * 2];
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s->filter[1][i] = filt[i * 2 + 1];
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}
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return 0;
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}
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static void convolve(AVFilterContext *ctx, AVFrame *in,
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int input_ch, int output_ch,
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int filter_ch, int tap_ch)
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{
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EarwaxContext *s = ctx->priv;
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int16_t *taps, *endin, *dst, *src;
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int len;
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if (!outsamples) {
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av_frame_free(&insamples);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(outsamples, insamples);
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taps = s->taps[tap_ch];
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dst = (int16_t *)s->frame[input_ch]->data[output_ch];
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src = (int16_t *)in->data[input_ch];
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taps = ((EarwaxContext *)inlink->dst->priv)->taps;
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out = (int16_t *)outsamples->data[0];
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in = (int16_t *)insamples ->data[0];
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len = FFMIN(NUMTAPS, 2*insamples->nb_samples);
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len = FFMIN(NUMTAPS, in->nb_samples);
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// copy part of new input and process with saved input
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memcpy(taps+NUMTAPS, in, len * sizeof(*taps));
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out = scalarproduct(taps, taps + len, out);
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memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
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dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
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// process current input
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if (2*insamples->nb_samples >= NUMTAPS ){
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endin = in + insamples->nb_samples * 2 - NUMTAPS;
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scalarproduct(in, endin, out);
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if (2*in->nb_samples >= NUMTAPS ){
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endin = src + in->nb_samples - NUMTAPS;
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scalarproduct(src, endin, s->filter[filter_ch], dst);
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// save part of input for next round
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memcpy(taps, endin, NUMTAPS * sizeof(*taps));
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} else
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memmove(taps, taps + 2*insamples->nb_samples, NUMTAPS * sizeof(*taps));
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} else {
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memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
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}
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}
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av_frame_free(&insamples);
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return ff_filter_frame(outlink, outsamples);
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static void mix(AVFilterContext *ctx, AVFrame *out,
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int output_ch, int f0, int f1, int i0, int i1)
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{
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EarwaxContext *s = ctx->priv;
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const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
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const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
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int16_t *dst = (int16_t *)out->data[output_ch];
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for (int n = 0; n < out->nb_samples; n++)
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dst[n] = av_clip_int16(srcl[n] + srcr[n]);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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EarwaxContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
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for (int ch = 0; ch < 2; ch++) {
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if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
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av_frame_free(&s->frame[ch]);
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s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!s->frame[ch]) {
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av_frame_free(&in);
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av_frame_free(&out);
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return AVERROR(ENOMEM);
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}
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}
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}
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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convolve(ctx, in, 0, 0, 0, 0);
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convolve(ctx, in, 0, 1, 1, 1);
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convolve(ctx, in, 1, 0, 0, 2);
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convolve(ctx, in, 1, 1, 1, 3);
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mix(ctx, out, 0, 0, 1, 1, 0);
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mix(ctx, out, 1, 0, 1, 0, 1);
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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EarwaxContext *s = ctx->priv;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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}
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static const AVFilterPad earwax_inputs[] = {
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@ -152,6 +218,7 @@ static const AVFilterPad earwax_inputs[] = {
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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@ -169,6 +236,7 @@ AVFilter ff_af_earwax = {
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.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
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.query_formats = query_formats,
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.priv_size = sizeof(EarwaxContext),
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.uninit = uninit,
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.inputs = earwax_inputs,
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.outputs = earwax_outputs,
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};
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@ -112,7 +112,7 @@ fate-filter-dcshift: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,dcshift
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FATE_AFILTER-$(call FILTERDEMDECENCMUX, EARWAX, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-earwax
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fate-filter-earwax: tests/data/asynth-44100-2.wav
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fate-filter-earwax: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
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fate-filter-earwax: CMD = framecrc -i $(SRC) -frames:a 20 -af earwax
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fate-filter-earwax: CMD = framecrc -i $(SRC) -frames:a 20 -af aresample,earwax,aresample
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FATE_AFILTER-$(call FILTERDEMDECENCMUX, EXTRASTEREO, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-extrastereo
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fate-filter-extrastereo: tests/data/asynth-44100-2.wav
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@ -4,23 +4,23 @@
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#sample_rate 0: 44100
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#channel_layout 0: 3
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#channel_layout_name 0: stereo
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0, 0, 0, 1024, 4096, 0x900af751
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0, 1024, 1024, 1024, 4096, 0xad570065
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0, 2048, 2048, 1024, 4096, 0x93d5f494
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0, 3072, 3072, 1024, 4096, 0x2c65ef7d
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0, 4096, 4096, 1024, 4096, 0xdc8af6d2
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0, 5120, 5120, 1024, 4096, 0x7ae00249
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0, 6144, 6144, 1024, 4096, 0xaab5fdd0
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0, 7168, 7168, 1024, 4096, 0x4373ef39
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0, 8192, 8192, 1024, 4096, 0x0756eb43
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0, 9216, 9216, 1024, 4096, 0x494d06e0
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0, 10240, 10240, 1024, 4096, 0x4393ffae
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0, 11264, 11264, 1024, 4096, 0x6972f97e
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0, 12288, 12288, 1024, 4096, 0xb834ea05
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0, 13312, 13312, 1024, 4096, 0x39b8f871
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0, 14336, 14336, 1024, 4096, 0xf032fccd
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0, 15360, 15360, 1024, 4096, 0xefcd0709
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0, 16384, 16384, 1024, 4096, 0x0590ebc0
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0, 17408, 17408, 1024, 4096, 0x2e75f264
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0, 18432, 18432, 1024, 4096, 0xbea1fd03
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0, 19456, 19456, 1024, 4096, 0x9bbe0434
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0, 0, 0, 1024, 4096, 0xb7e1f437
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0, 1024, 1024, 1024, 4096, 0xa031042a
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0, 2048, 2048, 1024, 4096, 0x9b72ed0f
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0, 3072, 3072, 1024, 4096, 0xff14ed33
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0, 4096, 4096, 1024, 4096, 0x96eef519
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0, 5120, 5120, 1024, 4096, 0x290d0ca0
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0, 6144, 6144, 1024, 4096, 0x0393fbf5
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0, 7168, 7168, 1024, 4096, 0xed89ef59
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0, 8192, 8192, 1024, 4096, 0xf664e969
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0, 9216, 9216, 1024, 4096, 0x261a05e4
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0, 10240, 10240, 1024, 4096, 0xc334ff5b
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0, 11264, 11264, 1024, 4096, 0x030ffa65
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0, 12288, 12288, 1024, 4096, 0xcfb4e835
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0, 13312, 13312, 1024, 4096, 0xd9adf7ff
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0, 14336, 14336, 1024, 4096, 0x5e9001ae
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0, 15360, 15360, 1024, 4096, 0xbfaf0174
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0, 16384, 16384, 1024, 4096, 0x8cf3f061
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0, 17408, 17408, 1024, 4096, 0x35ffece5
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0, 18432, 18432, 1024, 4096, 0x1de801e2
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0, 19456, 19456, 1024, 4096, 0xa1a40372
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