avfilter/af_anlmdn: stop using fifo and rewriting pts

libavfilter/af_anlmdn.c

@@ -21,12 +21,12 @@
 #include <float.h>
 
 #include "libavutil/avassert.h"
-#include "libavutil/audio_fifo.h"
 #include "libavutil/avstring.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "formats.h"
+#include "filters.h"
 
 #include "af_anlmdndsp.h"
 
@@ -50,14 +50,8 @@ typedef struct AudioNLMeansContext {
     int N;
     int H;
 
-    int offset;
-    AVFrame *in;
     AVFrame *cache;
-
+    AVFrame *window;
-    int64_t pts;
-
-    AVAudioFifo *fifo;
-    int eof_left;
 
     AudioNLMDNDSPContext dsp;
 } AudioNLMeansContext;
@@ -132,7 +126,6 @@ static int config_filter(AVFilterContext *ctx)
     AudioNLMeansContext *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
     int newK, newS, newH, newN;
-    AVFrame *new_in, *new_cache;
 
     newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
     newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
@@ -143,8 +136,11 @@ static int config_filter(AVFilterContext *ctx)
     av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
 
     if (!s->cache || s->cache->nb_samples < newS * 2) {
-        new_cache = ff_get_audio_buffer(outlink, newS * 2);
+        AVFrame *new_cache = ff_get_audio_buffer(outlink, newS * 2);
         if (new_cache) {
+            if (s->cache)
+                av_samples_copy(new_cache->extended_data, s->cache->extended_data, 0, 0,
+                                s->cache->nb_samples, new_cache->channels, new_cache->format);
             av_frame_free(&s->cache);
             s->cache = new_cache;
         } else {
@@ -154,6 +150,21 @@ static int config_filter(AVFilterContext *ctx)
     if (!s->cache)
         return AVERROR(ENOMEM);
 
+    if (!s->window || s->window->nb_samples < newN) {
+        AVFrame *new_window = ff_get_audio_buffer(outlink, newN);
+        if (new_window) {
+            if (s->window)
+                av_samples_copy(new_window->extended_data, s->window->extended_data, 0, 0,
+                                s->window->nb_samples, new_window->channels, new_window->format);
+            av_frame_free(&s->window);
+            s->window = new_window;
+        } else {
+            return AVERROR(ENOMEM);
+        }
+    }
+    if (!s->window)
+        return AVERROR(ENOMEM);
+
     s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
     for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
         float w = -i / s->pdiff_lut_scale;
@@ -161,18 +172,6 @@ static int config_filter(AVFilterContext *ctx)
         s->weight_lut[i] = expf(w);
     }
 
-    if (!s->in || s->in->nb_samples < newN) {
-        new_in = ff_get_audio_buffer(outlink, newN);
-        if (new_in) {
-            av_frame_free(&s->in);
-            s->in = new_in;
-        } else {
-            return AVERROR(ENOMEM);
-        }
-    }
-    if (!s->in)
-        return AVERROR(ENOMEM);
-
     s->K = newK;
     s->S = newS;
     s->H = newH;
@@ -187,21 +186,10 @@ static int config_output(AVFilterLink *outlink)
     AudioNLMeansContext *s = ctx->priv;
     int ret;
 
-    s->eof_left = -1;
-    s->pts = AV_NOPTS_VALUE;
-
     ret = config_filter(ctx);
     if (ret < 0)
         return ret;
 
-    s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
-    if (!s->fifo)
-        return AVERROR(ENOMEM);
-
-    ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
-    if (ret < 0)
-        return ret;
-
     ff_anlmdn_init(&s->dsp);
 
     return 0;
@@ -214,10 +202,10 @@ static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
     const int S = s->S;
     const int K = s->K;
     const int om = s->om;
-    const float *f = (const float *)(s->in->extended_data[ch]) + K;
+    const float *f = (const float *)(s->window->extended_data[ch]) + K;
     float *cache = (float *)s->cache->extended_data[ch];
     const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
-    float *dst = (float *)out->extended_data[ch] + s->offset;
+    float *dst = (float *)out->extended_data[ch];
     const float *const weight_lut = s->weight_lut;
     const float pdiff_lut_scale = s->pdiff_lut_scale;
     const float smooth = fminf(s->m, WEIGHT_LUT_SIZE / pdiff_lut_scale);
@@ -272,77 +260,56 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AudioNLMeansContext *s = ctx->priv;
-    AVFrame *out = NULL;
+    const int offset = s->N - s->H;
-    int available, wanted, ret;
+    AVFrame *out;
 
-    if (s->pts == AV_NOPTS_VALUE)
+    out = ff_get_audio_buffer(outlink, in->nb_samples);
-        s->pts = in->pts;
-
-    ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
-                              in->nb_samples);
-    av_frame_free(&in);
-
-    s->offset = 0;
-    available = av_audio_fifo_size(s->fifo);
-    wanted = (available / s->H) * s->H;
-
-    if (wanted >= s->H && available >= s->N) {
-        out = ff_get_audio_buffer(outlink, wanted);
     if (!out)
         return AVERROR(ENOMEM);
-    }
 
-    while (available >= s->N) {
+    for (int ch = 0; ch < in->channels; ch++) {
-        ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
+        float *src = (float *)s->window->extended_data[ch];
-        if (ret < 0)
+
-            break;
+        memmove(src, &src[s->H], offset * sizeof(float));
+        memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
+        memset(&src[offset + in->nb_samples], 0, (s->H - in->nb_samples) * sizeof(float));
+    }
 
     ff_filter_execute(ctx, filter_channel, out, NULL, inlink->channels);
 
-        av_audio_fifo_drain(s->fifo, s->H);
+    out->pts = in->pts;
-
+    av_frame_free(&in);
-        s->offset += s->H;
-        available -= s->H;
-    }
-
-    if (out) {
-        out->pts = s->pts;
-        out->nb_samples = s->offset;
-        if (s->eof_left >= 0) {
-            out->nb_samples = FFMIN(s->eof_left, s->offset);
-            s->eof_left -= out->nb_samples;
-        }
-        s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
-
     return ff_filter_frame(outlink, out);
-    }
-
-    return ret;
 }
 
-static int request_frame(AVFilterLink *outlink)
+static int activate(AVFilterContext *ctx)
 {
-    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
     AudioNLMeansContext *s = ctx->priv;
-    int ret;
+    AVFrame *in = NULL;
+    int ret = 0, status;
+    int64_t pts;
 
-    ret = ff_request_frame(ctx->inputs[0]);
+    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
-
-    if (ret == AVERROR_EOF && s->eof_left != 0) {
-        AVFrame *in;
-
-        if (s->eof_left < 0)
-            s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
-        if (s->eof_left <= 0)
-            return AVERROR_EOF;
-        in = ff_get_audio_buffer(outlink, s->H);
-        if (!in)
-            return AVERROR(ENOMEM);
-
-        return filter_frame(ctx->inputs[0], in);
-    }
 
+    ret = ff_inlink_consume_samples(inlink, s->H, s->H, &in);
+    if (ret < 0)
         return ret;
+
+    if (ret > 0) {
+        return filter_frame(inlink, in);
+    } else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+        ff_outlink_set_status(outlink, status, pts);
+        return 0;
+    } else {
+        if (ff_inlink_queued_samples(inlink) >= s->H) {
+            ff_filter_set_ready(ctx, 10);
+        } else if (ff_outlink_frame_wanted(outlink)) {
+            ff_inlink_request_frame(inlink);
+        }
+        return 0;
+    }
 }
 
 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
@@ -354,27 +321,21 @@ static int process_command(AVFilterContext *ctx, const char *cmd, const char *ar
     if (ret < 0)
         return ret;
 
-    ret = config_filter(ctx);
+    return config_filter(ctx);
-    if (ret < 0)
-        return ret;
-
-    return 0;
 }
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioNLMeansContext *s = ctx->priv;
 
-    av_audio_fifo_free(s->fifo);
-    av_frame_free(&s->in);
     av_frame_free(&s->cache);
+    av_frame_free(&s->window);
 }
 
 static const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
-        .filter_frame = filter_frame,
     },
 };
 
@@ -383,7 +344,6 @@ static const AVFilterPad outputs[] = {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
-        .request_frame = request_frame,
     },
 };
 
@@ -392,6 +352,7 @@ const AVFilter ff_af_anlmdn = {
     .description   = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
     .priv_size     = sizeof(AudioNLMeansContext),
     .priv_class    = &anlmdn_class,
+    .activate      = activate,
     .uninit        = uninit,
     FILTER_INPUTS(inputs),
     FILTER_OUTPUTS(outputs),