mirror of https://git.ffmpeg.org/ffmpeg.git
avfilter/af_afftdn: stop using fifo and rewritting pts
This commit is contained in:
Paul B Mahol
parent
c1735bb139
commit
644b6ed3ff
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@ -20,7 +20,6 @@
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#include <float.h>
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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@ -101,7 +100,6 @@ typedef struct AudioFFTDeNoiseContext {
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float last_noise_balance;
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int64_t block_count;
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int64_t pts;
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int channels;
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int sample_noise;
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int sample_noise_start;
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@ -124,6 +122,8 @@ typedef struct AudioFFTDeNoiseContext {
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DeNoiseChannel *dnch;
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AVFrame *winframe;
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double max_gain;
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double max_var;
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double gain_scale;
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@ -138,8 +138,6 @@ typedef struct AudioFFTDeNoiseContext {
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double vector_b[5];
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double matrix_b[75];
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double matrix_c[75];
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AVAudioFifo *fifo;
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} AudioFFTDeNoiseContext;
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#define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
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@ -620,7 +618,6 @@ static int config_input(AVFilterLink *inlink)
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if (!s->dnch)
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return AVERROR(ENOMEM);
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s->pts = AV_NOPTS_VALUE;
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s->channels = inlink->channels;
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s->sample_rate = inlink->sample_rate;
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s->sample_advance = s->sample_rate / 80;
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@ -831,6 +828,10 @@ static int config_input(AVFilterLink *inlink)
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}
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}
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s->winframe = ff_get_audio_buffer(inlink, s->window_length);
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if (!s->winframe)
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return AVERROR(ENOMEM);
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wscale = sqrt(16.0 / (9.0 * s->fft_length));
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sum = 0.0;
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for (int i = 0; i < s->window_length; i++) {
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@ -857,10 +858,6 @@ static int config_input(AVFilterLink *inlink)
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}
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s->noise_band_count = s->noise_band_edge[16];
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s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->fft_length);
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if (!s->fifo)
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return AVERROR(ENOMEM);
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return 0;
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}
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@ -1160,23 +1157,24 @@ static void get_auto_noise_levels(AudioFFTDeNoiseContext *s,
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}
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}
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static int output_frame(AVFilterLink *inlink)
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static int output_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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AudioFFTDeNoiseContext *s = ctx->priv;
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const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode;
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AVFrame *out = NULL, *in = NULL;
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const int offset = s->window_length - s->sample_advance;
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AVFrame *out = NULL;
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ThreadData td;
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int ret = 0;
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in = ff_get_audio_buffer(outlink, s->window_length);
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if (!in)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < s->channels; ch++) {
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float *src = (float *)s->winframe->extended_data[ch];
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ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length);
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if (ret < 0)
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goto end;
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memmove(src, &src[s->sample_advance], offset * sizeof(float));
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memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
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memset(&src[offset + in->nb_samples], 0, (s->sample_advance - in->nb_samples) * sizeof(float));
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}
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if (s->track_noise) {
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for (int ch = 0; ch < inlink->channels; ch++) {
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@ -1205,7 +1203,7 @@ static int output_frame(AVFilterLink *inlink)
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for (int ch = 0; ch < inlink->channels; ch++) {
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DeNoiseChannel *dnch = &s->dnch[ch];
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sample_noise_block(s, dnch, in, ch);
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sample_noise_block(s, dnch, s->winframe, ch);
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}
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}
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@ -1223,11 +1221,11 @@ static int output_frame(AVFilterLink *inlink)
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}
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s->block_count++;
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td.in = in;
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td.in = s->winframe;
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ff_filter_execute(ctx, filter_channel, &td, NULL,
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FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
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out = ff_get_audio_buffer(outlink, s->sample_advance);
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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ret = AVERROR(ENOMEM);
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goto end;
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@ -1236,20 +1234,20 @@ static int output_frame(AVFilterLink *inlink)
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for (int ch = 0; ch < inlink->channels; ch++) {
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DeNoiseChannel *dnch = &s->dnch[ch];
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double *src = dnch->out_samples;
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float *orig = (float *)in->extended_data[ch];
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const float *orig = (const float *)s->winframe->extended_data[ch];
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float *dst = (float *)out->extended_data[ch];
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switch (output_mode) {
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case IN_MODE:
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for (int m = 0; m < s->sample_advance; m++)
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for (int m = 0; m < out->nb_samples; m++)
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dst[m] = orig[m];
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break;
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case OUT_MODE:
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for (int m = 0; m < s->sample_advance; m++)
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for (int m = 0; m < out->nb_samples; m++)
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dst[m] = src[m];
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break;
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case NOISE_MODE:
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for (int m = 0; m < s->sample_advance; m++)
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for (int m = 0; m < out->nb_samples; m++)
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dst[m] = orig[m] - src[m];
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break;
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default:
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@ -1261,13 +1259,11 @@ static int output_frame(AVFilterLink *inlink)
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memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
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}
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av_audio_fifo_drain(s->fifo, s->sample_advance);
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out->pts = in->pts;
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out->pts = s->pts;
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ret = ff_filter_frame(outlink, out);
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if (ret < 0)
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goto end;
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s->pts += av_rescale_q(s->sample_advance, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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end:
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av_frame_free(&in);
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@ -1279,34 +1275,19 @@ static int activate(AVFilterContext *ctx)
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioFFTDeNoiseContext *s = ctx->priv;
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AVFrame *frame = NULL;
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AVFrame *in = NULL;
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int ret;
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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ret = ff_inlink_consume_frame(inlink, &frame);
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ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
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if (ret < 0)
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return ret;
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if (ret > 0) {
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if (s->pts == AV_NOPTS_VALUE)
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s->pts = frame->pts;
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ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
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av_frame_free(&frame);
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if (ret < 0)
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return ret;
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}
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if (av_audio_fifo_size(s->fifo) >= s->window_length)
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return output_frame(inlink);
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if (ret > 0)
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return output_frame(inlink, in);
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FF_FILTER_FORWARD_STATUS(inlink, outlink);
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if (ff_outlink_frame_wanted(outlink) &&
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av_audio_fifo_size(s->fifo) < s->window_length) {
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ff_inlink_request_frame(inlink);
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return 0;
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}
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return FFERROR_NOT_READY;
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}
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@ -1319,6 +1300,7 @@ static av_cold void uninit(AVFilterContext *ctx)
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av_freep(&s->bin2band);
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av_freep(&s->band_alpha);
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av_freep(&s->band_beta);
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av_frame_free(&s->winframe);
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if (s->dnch) {
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for (int ch = 0; ch < s->channels; ch++) {
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@ -1343,8 +1325,6 @@ static av_cold void uninit(AVFilterContext *ctx)
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}
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av_freep(&s->dnch);
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}
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av_audio_fifo_free(s->fifo);
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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