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authorPaul B Mahol <onemda@gmail.com>2022-03-09 21:39:46 +0100
committerPaul B Mahol <onemda@gmail.com>2022-03-09 22:08:36 +0100
commitc71b76e14196ece59ef1f3f0216878e25de3b46a (patch)
treee8278e14f11b31705bd76e61762ca183a400fa2e
parent41cae501b7f2089579bf4bd8f938520a37dede55 (diff)
downloadffmpeg-c71b76e14196ece59ef1f3f0216878e25de3b46a.tar.gz
avfilter/af_anlmdn: stop using fifo and rewriting pts
-rw-r--r--libavfilter/af_anlmdn.c163
1 files changed, 62 insertions, 101 deletions
diff --git a/libavfilter/af_anlmdn.c b/libavfilter/af_anlmdn.c
index a2c42393b6..9df388b727 100644
--- a/libavfilter/af_anlmdn.c
+++ b/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;
-
- int64_t pts;
-
- AVAudioFifo *fifo;
- int eof_left;
+ AVFrame *window;
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;
- int available, wanted, ret;
-
- if (s->pts == AV_NOPTS_VALUE)
- 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;
+ const int offset = s->N - s->H;
+ AVFrame *out;
- if (wanted >= s->H && available >= s->N) {
- out = ff_get_audio_buffer(outlink, wanted);
- if (!out)
- return AVERROR(ENOMEM);
- }
-
- while (available >= s->N) {
- ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
- if (ret < 0)
- break;
-
- ff_filter_execute(ctx, filter_channel, out, NULL, inlink->channels);
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
- av_audio_fifo_drain(s->fifo, s->H);
+ for (int ch = 0; ch < in->channels; ch++) {
+ float *src = (float *)s->window->extended_data[ch];
- s->offset += s->H;
- available -= s->H;
+ 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));
}
- 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);
- }
+ ff_filter_execute(ctx, filter_channel, out, NULL, inlink->channels);
- return ret;
+ out->pts = in->pts;
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
}
-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;
-
- ret = ff_request_frame(ctx->inputs[0]);
+ AVFrame *in = NULL;
+ int ret = 0, status;
+ int64_t pts;
- if (ret == AVERROR_EOF && s->eof_left != 0) {
- AVFrame *in;
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
- 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);
+ ret = ff_inlink_consume_samples(inlink, s->H, s->H, &in);
+ if (ret < 0)
+ return ret;
- return filter_frame(ctx->inputs[0], in);
+ 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;
}
-
- return ret;
}
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);
- if (ret < 0)
- return ret;
-
- return 0;
+ return config_filter(ctx);
}
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),