diff mbox series

[FFmpeg-devel] avfilter/af_earwax: fix filter behavior

Message ID 20201204170345.11668-1-onemda@gmail.com
State Accepted
Headers show
Series [FFmpeg-devel] avfilter/af_earwax: fix filter behavior
Related show

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Context Check Description
andriy/x86_make success Make finished
andriy/x86_make_fate success Make fate finished

Commit Message

Paul B Mahol Dec. 4, 2020, 5:03 p.m. UTC
Previous filter output was incorrect. New one actually follows
graph in comments described on side of filter taps.

Signed-off-by: Paul B Mahol <onemda@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(-)

Comments

Paul B Mahol Dec. 7, 2020, 12:17 p.m. UTC | #1
Will apply soon with few leaks on the allocation error fixed.

On Fri, Dec 4, 2020 at 6:03 PM Paul B Mahol <onemda@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 <onemda@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
>
>
diff mbox series

Patch

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
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