Will apply soon with few leaks on the allocation error fixed.
On Fri, Dec 4, 2020 at 6:03 PM Paul B Mahol <one...@gmail.com> wrote:
> Previous filter output was incorrect. New one actually follows
> graph in comments described on side of filter taps.
>
> Signed-off-by: Paul B Mahol <one...@gmail.com>
> ---
> libavfilter/af_earwax.c | 119 +++++++++++++++++++++++++++--------
> tests/fate/filter-audio.mak | 2 +-
> tests/ref/fate/filter-earwax | 40 ++++++------
> 3 files changed, 113 insertions(+), 48 deletions(-)
>
> diff --git a/libavfilter/af_earwax.c b/libavfilter/af_earwax.c
> index cdd2b4fc49..951aaccb5d 100644
> --- a/libavfilter/af_earwax.c
> +++ b/libavfilter/af_earwax.c
> @@ -34,9 +34,9 @@
> #include "audio.h"
> #include "formats.h"
>
> -#define NUMTAPS 64
> +#define NUMTAPS 32
>
> -static const int8_t filt[NUMTAPS] = {
> +static const int8_t filt[NUMTAPS * 2] = {
> /* 30° 330° */
> 4, -6, /* 32 tap stereo FIR filter. */
> 4, -11, /* One side filters as if the */
> @@ -72,7 +72,10 @@ static const int8_t filt[NUMTAPS] = {
> 4, 0};
>
> typedef struct EarwaxContext {
> - int16_t taps[NUMTAPS * 2];
> + int16_t filter[2][NUMTAPS];
> + int16_t taps[4][NUMTAPS * 2];
> +
> + AVFrame *frame[2];
> } EarwaxContext;
>
> static int query_formats(AVFilterContext *ctx)
> @@ -83,7 +86,7 @@ static int query_formats(AVFilterContext *ctx)
> AVFilterFormats *formats = NULL;
> AVFilterChannelLayouts *layout = NULL;
>
> - if ((ret = ff_add_format (&formats,
> AV_SAMPLE_FMT_S16 )) < 0 ||
> + if ((ret = ff_add_format (&formats,
> AV_SAMPLE_FMT_S16P )) < 0 ||
> (ret = ff_set_common_formats (ctx , formats
> )) < 0 ||
> (ret = ff_add_channel_layout (&layout ,
> AV_CH_LAYOUT_STEREO )) < 0 ||
> (ret = ff_set_common_channel_layouts (ctx , layout
> )) < 0 ||
> @@ -94,7 +97,8 @@ static int query_formats(AVFilterContext *ctx)
> }
>
> //FIXME: replace with DSPContext.scalarproduct_int16
> -static inline int16_t *scalarproduct(const int16_t *in, const int16_t
> *endin, int16_t *out)
> +static inline int16_t *scalarproduct(const int16_t *in, const int16_t
> *endin,
> + const int16_t *filt, int16_t *out)
> {
> int32_t sample;
> int16_t j;
> @@ -111,40 +115,99 @@ static inline int16_t *scalarproduct(const int16_t
> *in, const int16_t *endin, in
> return out;
> }
>
> -static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
> +static int config_input(AVFilterLink *inlink)
> {
> - AVFilterLink *outlink = inlink->dst->outputs[0];
> - int16_t *taps, *endin, *in, *out;
> - AVFrame *outsamples = ff_get_audio_buffer(outlink,
> insamples->nb_samples);
> - int len;
> + EarwaxContext *s = inlink->dst->priv;
>
> - if (!outsamples) {
> - av_frame_free(&insamples);
> - return AVERROR(ENOMEM);
> + for (int i = 0; i < NUMTAPS; i++) {
> + s->filter[0][i] = filt[i * 2];
> + s->filter[1][i] = filt[i * 2 + 1];
> }
> - av_frame_copy_props(outsamples, insamples);
>
> - taps = ((EarwaxContext *)inlink->dst->priv)->taps;
> - out = (int16_t *)outsamples->data[0];
> - in = (int16_t *)insamples ->data[0];
> + return 0;
> +}
> +
> +static void convolve(AVFilterContext *ctx, AVFrame *in,
> + int input_ch, int output_ch,
> + int filter_ch, int tap_ch)
> +{
> + EarwaxContext *s = ctx->priv;
> + int16_t *taps, *endin, *dst, *src;
> + int len;
> +
> + taps = s->taps[tap_ch];
> + dst = (int16_t *)s->frame[input_ch]->data[output_ch];
> + src = (int16_t *)in->data[input_ch];
>
> - len = FFMIN(NUMTAPS, 2*insamples->nb_samples);
> + len = FFMIN(NUMTAPS, in->nb_samples);
> // copy part of new input and process with saved input
> - memcpy(taps+NUMTAPS, in, len * sizeof(*taps));
> - out = scalarproduct(taps, taps + len, out);
> + memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
> + dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
>
> // process current input
> - if (2*insamples->nb_samples >= NUMTAPS ){
> - endin = in + insamples->nb_samples * 2 - NUMTAPS;
> - scalarproduct(in, endin, out);
> + if (2*in->nb_samples >= NUMTAPS ){
> + endin = src + in->nb_samples - NUMTAPS;
> + scalarproduct(src, endin, s->filter[filter_ch], dst);
>
> // save part of input for next round
> memcpy(taps, endin, NUMTAPS * sizeof(*taps));
> - } else
> - memmove(taps, taps + 2*insamples->nb_samples, NUMTAPS *
> sizeof(*taps));
> + } else {
> + memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
> + }
> +}
> +
> +static void mix(AVFilterContext *ctx, AVFrame *out,
> + int output_ch, int f0, int f1, int i0, int i1)
> +{
> + EarwaxContext *s = ctx->priv;
> + const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
> + const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
> + int16_t *dst = (int16_t *)out->data[output_ch];
> +
> + for (int n = 0; n < out->nb_samples; n++)
> + dst[n] = srcl[n] + srcr[n];
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + EarwaxContext *s = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> + AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
> +
> + for (int ch = 0; ch < 2; ch++) {
> + if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
> + av_frame_free(&s->frame[ch]);
> + s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
> + if (!s->frame[ch])
> + return AVERROR(ENOMEM);
> + }
> + }
> +
> + if (!out) {
> + av_frame_free(&in);
> + return AVERROR(ENOMEM);
> + }
> + av_frame_copy_props(out, in);
> +
> + convolve(ctx, in, 0, 0, 0, 0);
> + convolve(ctx, in, 0, 1, 1, 1);
> + convolve(ctx, in, 1, 0, 0, 2);
> + convolve(ctx, in, 1, 1, 1, 3);
> +
> + mix(ctx, out, 0, 0, 1, 1, 0);
> + mix(ctx, out, 1, 0, 1, 0, 1);
> +
> + av_frame_free(&in);
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + EarwaxContext *s = ctx->priv;
>
> - av_frame_free(&insamples);
> - return ff_filter_frame(outlink, outsamples);
> + av_frame_free(&s->frame[0]);
> + av_frame_free(&s->frame[1]);
> }
>
> static const AVFilterPad earwax_inputs[] = {
> @@ -152,6 +215,7 @@ static const AVFilterPad earwax_inputs[] = {
> .name = "default",
> .type = AVMEDIA_TYPE_AUDIO,
> .filter_frame = filter_frame,
> + .config_props = config_input,
> },
> { NULL }
> };
> @@ -169,6 +233,7 @@ AVFilter ff_af_earwax = {
> .description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
> .query_formats = query_formats,
> .priv_size = sizeof(EarwaxContext),
> + .uninit = uninit,
> .inputs = earwax_inputs,
> .outputs = earwax_outputs,
> };
> diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
> index ee1a9dfc97..8b38ee5e75 100644
> --- a/tests/fate/filter-audio.mak
> +++ b/tests/fate/filter-audio.mak
> @@ -112,7 +112,7 @@ fate-filter-dcshift: CMD = framecrc -i $(SRC)
> -frames:a 20 -af aresample,dcshift
> FATE_AFILTER-$(call FILTERDEMDECENCMUX, EARWAX, WAV, PCM_S16LE,
> PCM_S16LE, WAV) += fate-filter-earwax
> fate-filter-earwax: tests/data/asynth-44100-2.wav
> fate-filter-earwax: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
> -fate-filter-earwax: CMD = framecrc -i $(SRC) -frames:a 20 -af earwax
> +fate-filter-earwax: CMD = framecrc -i $(SRC) -frames:a 20 -af
> aresample,earwax,aresample
>
> FATE_AFILTER-$(call FILTERDEMDECENCMUX, EXTRASTEREO, WAV, PCM_S16LE,
> PCM_S16LE, WAV) += fate-filter-extrastereo
> fate-filter-extrastereo: tests/data/asynth-44100-2.wav
> diff --git a/tests/ref/fate/filter-earwax b/tests/ref/fate/filter-earwax
> index 855f579cac..b079ea404b 100644
> --- a/tests/ref/fate/filter-earwax
> +++ b/tests/ref/fate/filter-earwax
> @@ -4,23 +4,23 @@
> #sample_rate 0: 44100
> #channel_layout 0: 3
> #channel_layout_name 0: stereo
> -0, 0, 0, 1024, 4096, 0x900af751
> -0, 1024, 1024, 1024, 4096, 0xad570065
> -0, 2048, 2048, 1024, 4096, 0x93d5f494
> -0, 3072, 3072, 1024, 4096, 0x2c65ef7d
> -0, 4096, 4096, 1024, 4096, 0xdc8af6d2
> -0, 5120, 5120, 1024, 4096, 0x7ae00249
> -0, 6144, 6144, 1024, 4096, 0xaab5fdd0
> -0, 7168, 7168, 1024, 4096, 0x4373ef39
> -0, 8192, 8192, 1024, 4096, 0x0756eb43
> -0, 9216, 9216, 1024, 4096, 0x494d06e0
> -0, 10240, 10240, 1024, 4096, 0x4393ffae
> -0, 11264, 11264, 1024, 4096, 0x6972f97e
> -0, 12288, 12288, 1024, 4096, 0xb834ea05
> -0, 13312, 13312, 1024, 4096, 0x39b8f871
> -0, 14336, 14336, 1024, 4096, 0xf032fccd
> -0, 15360, 15360, 1024, 4096, 0xefcd0709
> -0, 16384, 16384, 1024, 4096, 0x0590ebc0
> -0, 17408, 17408, 1024, 4096, 0x2e75f264
> -0, 18432, 18432, 1024, 4096, 0xbea1fd03
> -0, 19456, 19456, 1024, 4096, 0x9bbe0434
> +0, 0, 0, 1024, 4096, 0x76eff65b
> +0, 1024, 1024, 1024, 4096, 0x2621ffb7
> +0, 2048, 2048, 1024, 4096, 0x40bfeb89
> +0, 3072, 3072, 1024, 4096, 0xd530f217
> +0, 4096, 4096, 1024, 4096, 0x34f6f5fb
> +0, 5120, 5120, 1024, 4096, 0x4aaa04c6
> +0, 6144, 6144, 1024, 4096, 0x73d5fb9f
> +0, 7168, 7168, 1024, 4096, 0x5726ee2b
> +0, 8192, 8192, 1024, 4096, 0x8b40ec57
> +0, 9216, 9216, 1024, 4096, 0xe5c3052a
> +0, 10240, 10240, 1024, 4096, 0xd300f85d
> +0, 11264, 11264, 1024, 4096, 0xdedaf89d
> +0, 12288, 12288, 1024, 4096, 0xeaa2ea0f
> +0, 13312, 13312, 1024, 4096, 0xf014fba1
> +0, 14336, 14336, 1024, 4096, 0x59f2fcdd
> +0, 15360, 15360, 1024, 4096, 0x1cbc00b0
> +0, 16384, 16384, 1024, 4096, 0x1423ee75
> +0, 17408, 17408, 1024, 4096, 0xdab7ef27
> +0, 18432, 18432, 1024, 4096, 0xdfa20160
> +0, 19456, 19456, 1024, 4096, 0xe435fca9
> --
> 2.17.1
>
>
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