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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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int len; |
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EarwaxContext *s = inlink->dst->priv; |
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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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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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av_frame_copy_props(outsamples, insamples); |
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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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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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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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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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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(&insamples); |
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return ff_filter_frame(outlink, outsamples); |
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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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}; |