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author | Paul B Mahol <onemda@gmail.com> | 2018-10-28 14:27:32 +0100 |
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committer | Paul B Mahol <onemda@gmail.com> | 2018-10-28 14:31:03 +0100 |
commit | bb54c0ae719ef768463e78af82366e73c79e1e50 (patch) | |
tree | d2cb57de98b60fd95219f97f6562c3488ca103b5 /libavfilter | |
parent | bdfd2e3c79dd3d0a75033cad662a78fa0a04304f (diff) | |
download | ffmpeg-bb54c0ae719ef768463e78af82366e73c79e1e50.tar.gz |
avfilter/af_afftdn: switch to activate
Diffstat (limited to 'libavfilter')
-rw-r--r-- | libavfilter/af_afftdn.c | 203 |
1 files changed, 116 insertions, 87 deletions
diff --git a/libavfilter/af_afftdn.c b/libavfilter/af_afftdn.c index fbcb0f18d5..77e5406d1d 100644 --- a/libavfilter/af_afftdn.c +++ b/libavfilter/af_afftdn.c @@ -28,6 +28,7 @@ #include "avfilter.h" #include "audio.h" #include "formats.h" +#include "filters.h" #define C (M_LN10 * 0.1) #define RATIO 0.98 @@ -1153,7 +1154,7 @@ static void get_auto_noise_levels(AudioFFTDeNoiseContext *s, } } -static int filter_frame(AVFilterLink *inlink, AVFrame *frame) +static int output_frame(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; @@ -1162,117 +1163,145 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame) ThreadData td; int ret = 0; - if (s->pts == AV_NOPTS_VALUE) - s->pts = frame->pts; + if (!in) { + in = ff_get_audio_buffer(outlink, s->window_length); + if (!in) + return AVERROR(ENOMEM); + } - ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples); - av_frame_free(&frame); + ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length); if (ret < 0) return ret; - while (av_audio_fifo_size(s->fifo) >= s->window_length) { - if (!in) { - in = ff_get_audio_buffer(outlink, s->window_length); - if (!in) - return AVERROR(ENOMEM); - } + if (s->track_noise) { + for (int ch = 0; ch < inlink->channels; ch++) { + DeNoiseChannel *dnch = &s->dnch[ch]; + double levels[15]; - ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length); - if (ret < 0) - break; + get_auto_noise_levels(s, dnch, levels); + set_noise_profile(s, dnch, levels, 0); + } - if (s->track_noise) { - for (int ch = 0; ch < inlink->channels; ch++) { - DeNoiseChannel *dnch = &s->dnch[ch]; - double levels[15]; + if (s->noise_floor != s->last_noise_floor) + set_parameters(s); + } - get_auto_noise_levels(s, dnch, levels); - set_noise_profile(s, dnch, levels, 0); - } + if (s->sample_noise_start) { + for (int ch = 0; ch < inlink->channels; ch++) { + DeNoiseChannel *dnch = &s->dnch[ch]; - if (s->noise_floor != s->last_noise_floor) - set_parameters(s); + init_sample_noise(dnch); } + s->sample_noise_start = 0; + s->sample_noise = 1; + } - if (s->sample_noise_start) { - for (int ch = 0; ch < inlink->channels; ch++) { - DeNoiseChannel *dnch = &s->dnch[ch]; + if (s->sample_noise) { + for (int ch = 0; ch < inlink->channels; ch++) { + DeNoiseChannel *dnch = &s->dnch[ch]; - init_sample_noise(dnch); - } - s->sample_noise_start = 0; - s->sample_noise = 1; + sample_noise_block(s, dnch, in, ch); } + } - if (s->sample_noise) { - for (int ch = 0; ch < inlink->channels; ch++) { - DeNoiseChannel *dnch = &s->dnch[ch]; + if (s->sample_noise_end) { + for (int ch = 0; ch < inlink->channels; ch++) { + DeNoiseChannel *dnch = &s->dnch[ch]; + double sample_noise[15]; - sample_noise_block(s, dnch, in, ch); - } + finish_sample_noise(s, dnch, sample_noise); + set_noise_profile(s, dnch, sample_noise, 1); + set_band_parameters(s, dnch); } + s->sample_noise = 0; + s->sample_noise_end = 0; + } - if (s->sample_noise_end) { - for (int ch = 0; ch < inlink->channels; ch++) { - DeNoiseChannel *dnch = &s->dnch[ch]; - double sample_noise[15]; - - finish_sample_noise(s, dnch, sample_noise); - set_noise_profile(s, dnch, sample_noise, 1); - set_band_parameters(s, dnch); - } - s->sample_noise = 0; - s->sample_noise_end = 0; - } + s->block_count++; + td.in = in; + ctx->internal->execute(ctx, filter_channel, &td, NULL, + FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); - s->block_count++; - td.in = in; - ctx->internal->execute(ctx, filter_channel, &td, NULL, - FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); + out = ff_get_audio_buffer(outlink, s->sample_advance); + if (!out) { + ret = AVERROR(ENOMEM); + goto end; + } - out = ff_get_audio_buffer(outlink, s->sample_advance); - if (!out) { - ret = AVERROR(ENOMEM); + for (int ch = 0; ch < inlink->channels; ch++) { + DeNoiseChannel *dnch = &s->dnch[ch]; + double *src = dnch->out_samples; + float *orig = (float *)in->extended_data[ch]; + float *dst = (float *)out->extended_data[ch]; + + switch (s->output_mode) { + case IN_MODE: + for (int m = 0; m < s->sample_advance; m++) + dst[m] = orig[m]; + break; + case OUT_MODE: + for (int m = 0; m < s->sample_advance; m++) + dst[m] = src[m]; + break; + case NOISE_MODE: + for (int m = 0; m < s->sample_advance; m++) + dst[m] = orig[m] - src[m]; break; + default: + return AVERROR_BUG; } + memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src)); + memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src)); + } - for (int ch = 0; ch < inlink->channels; ch++) { - DeNoiseChannel *dnch = &s->dnch[ch]; - double *src = dnch->out_samples; - float *orig = (float *)in->extended_data[ch]; - float *dst = (float *)out->extended_data[ch]; - - switch (s->output_mode) { - case IN_MODE: - for (int m = 0; m < s->sample_advance; m++) - dst[m] = orig[m]; - break; - case OUT_MODE: - for (int m = 0; m < s->sample_advance; m++) - dst[m] = src[m]; - break; - case NOISE_MODE: - for (int m = 0; m < s->sample_advance; m++) - dst[m] = orig[m] - src[m]; - break; - default: - return AVERROR_BUG; - } - memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src)); - memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src)); - } + av_audio_fifo_drain(s->fifo, s->sample_advance); + + out->pts = s->pts; + ret = ff_filter_frame(outlink, out); + if (ret < 0) + goto end; + s->pts += s->sample_advance; +end: + av_frame_free(&in); - av_audio_fifo_drain(s->fifo, s->sample_advance); + return ret; +} - out->pts = s->pts; - ret = ff_filter_frame(outlink, out); +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + AudioFFTDeNoiseContext *s = ctx->priv; + AVFrame *frame = NULL; + int ret; + + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); + + ret = ff_inlink_consume_frame(inlink, &frame); + if (ret < 0) + return ret; + + if (ret > 0) { + if (s->pts == AV_NOPTS_VALUE) + s->pts = frame->pts; + + ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples); + av_frame_free(&frame); if (ret < 0) - break; - s->pts += s->sample_advance; + return ret; } - av_frame_free(&in); - return ret; + if (av_audio_fifo_size(s->fifo) >= s->window_length) + return output_frame(inlink); + + FF_FILTER_FORWARD_STATUS(inlink, outlink); + if (ff_outlink_frame_wanted(outlink) && + av_audio_fifo_size(s->fifo) < s->window_length) { + ff_inlink_request_frame(inlink); + return 0; + } + + return FFERROR_NOT_READY; } static av_cold void uninit(AVFilterContext *ctx) @@ -1393,7 +1422,6 @@ static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, .config_props = config_input, }, { NULL } @@ -1413,6 +1441,7 @@ AVFilter ff_af_afftdn = { .query_formats = query_formats, .priv_size = sizeof(AudioFFTDeNoiseContext), .priv_class = &afftdn_class, + .activate = activate, .uninit = uninit, .inputs = inputs, .outputs = outputs, |