Paul B Mahol (2018-10-03): > Signed-off-by: Paul B Mahol <one...@gmail.com> > --- > libavfilter/af_asetnsamples.c | 156 +++++++++------------------------- > 1 file changed, 42 insertions(+), 114 deletions(-) > > diff --git a/libavfilter/af_asetnsamples.c b/libavfilter/af_asetnsamples.c > index ecb76e64db..6efa6f3f69 100644 > --- a/libavfilter/af_asetnsamples.c > +++ b/libavfilter/af_asetnsamples.c > @@ -24,20 +24,18 @@ > * Filter that changes number of samples on single output operation > */ > > -#include "libavutil/audio_fifo.h" > #include "libavutil/avassert.h" > #include "libavutil/channel_layout.h" > #include "libavutil/opt.h" > #include "avfilter.h" > #include "audio.h" > +#include "filters.h" > #include "internal.h" > #include "formats.h" > > typedef struct ASNSContext { > const AVClass *class; > int nb_out_samples; ///< how many samples to output > - AVAudioFifo *fifo; ///< samples are queued here > - int64_t next_out_pts; > int pad; > } ASNSContext; > > @@ -54,134 +52,65 @@ static const AVOption asetnsamples_options[] = { > > AVFILTER_DEFINE_CLASS(asetnsamples); > > -static av_cold int init(AVFilterContext *ctx) > +static int activate(AVFilterContext *ctx) > { > - ASNSContext *asns = ctx->priv; > - > - asns->next_out_pts = AV_NOPTS_VALUE; > - av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", > asns->nb_out_samples, asns->pad); > - > - return 0; > -} > - > -static av_cold void uninit(AVFilterContext *ctx) > -{ > - ASNSContext *asns = ctx->priv; > - av_audio_fifo_free(asns->fifo); > -} > - > -static int config_props_output(AVFilterLink *outlink) > -{ > - ASNSContext *asns = outlink->src->priv; > - > - asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, > asns->nb_out_samples); > - if (!asns->fifo) > - return AVERROR(ENOMEM); > - > - return 0; > -} > - > -static int push_samples(AVFilterLink *outlink) > -{ > - ASNSContext *asns = outlink->src->priv; > - AVFrame *outsamples = NULL; > - int ret, nb_out_samples, nb_pad_samples; > - > - if (asns->pad) { > - nb_out_samples = av_audio_fifo_size(asns->fifo) ? > asns->nb_out_samples : 0; > - nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, > av_audio_fifo_size(asns->fifo)); > - } else { > - nb_out_samples = FFMIN(asns->nb_out_samples, > av_audio_fifo_size(asns->fifo)); > - nb_pad_samples = 0; > - } > - > - if (!nb_out_samples) > - return 0; > - > - outsamples = ff_get_audio_buffer(outlink, nb_out_samples); > - if (!outsamples) > - return AVERROR(ENOMEM); > - > - av_audio_fifo_read(asns->fifo, > - (void **)outsamples->extended_data, nb_out_samples); > - > - if (nb_pad_samples) > - av_samples_set_silence(outsamples->extended_data, nb_out_samples - > nb_pad_samples, > - nb_pad_samples, outlink->channels, > - outlink->format); > - outsamples->nb_samples = nb_out_samples; > - outsamples->channel_layout = outlink->channel_layout; > - outsamples->sample_rate = outlink->sample_rate; > - outsamples->pts = asns->next_out_pts; > - > - if (asns->next_out_pts != AV_NOPTS_VALUE) > - asns->next_out_pts += av_rescale_q(nb_out_samples, (AVRational){1, > outlink->sample_rate}, outlink->time_base); > - > - ret = ff_filter_frame(outlink, outsamples); > - if (ret < 0) > - return ret; > - return nb_out_samples; > -} > - > -static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) > -{ > - AVFilterContext *ctx = inlink->dst; > - ASNSContext *asns = ctx->priv; > + AVFilterLink *inlink = ctx->inputs[0]; > AVFilterLink *outlink = ctx->outputs[0]; > + ASNSContext *s = ctx->priv; > + AVFrame *frame = NULL; > int ret; > - int nb_samples = insamples->nb_samples; > - > - if (av_audio_fifo_space(asns->fifo) < nb_samples) { > - av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio > fifo\n", nb_samples); > - ret = av_audio_fifo_realloc(asns->fifo, > av_audio_fifo_size(asns->fifo) + nb_samples); > - if (ret < 0) { > - av_log(ctx, AV_LOG_ERROR, > - "Stretching audio fifo failed, discarded %d samples\n", > nb_samples); > - return -1; > - } > - } > - ret = av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, > nb_samples); > - if (ret > 0 && asns->next_out_pts == AV_NOPTS_VALUE) > - asns->next_out_pts = insamples->pts; > - av_frame_free(&insamples); > > - if (ret < 0) > - return ret; > + FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink); > > - while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples) > - push_samples(outlink); > - return 0; > -}
> + if (ff_inlink_queued_samples(inlink) >= s->nb_out_samples) { This test is not needed, just check the return value of ff_inlink_consume_samples(). > + ret = ff_inlink_consume_samples(inlink, s->nb_out_samples, > s->nb_out_samples, &frame); > + if (ret > 0) > + ret = ff_filter_frame(outlink, frame); > + return ret; I do not like that style, I prefer if the exceptional case is the one in the condition. > + } > > -static int request_frame(AVFilterLink *outlink) > -{ > - AVFilterLink *inlink = outlink->src->inputs[0]; > - int ret; > + if (ff_outlink_get_status(inlink) == AVERROR_EOF) { > + AVFrame *pad_frame; > + > + ret = ff_inlink_consume_samples(inlink, s->nb_out_samples, > s->nb_out_samples, &frame); This special case is not needed: ff_inlink_consume_samples() will return a smaller frame only at the end, so just pad it in the normal case. > + if (ret > 0 && s->pad && frame->nb_samples < s->nb_out_samples) { > + pad_frame = ff_get_audio_buffer(outlink, s->nb_out_samples); > + if (!pad_frame) > + return AVERROR(ENOMEM); > + > + av_samples_copy(pad_frame->extended_data, frame->extended_data, > + 0, 0, frame->nb_samples, frame->channels, > frame->format); > + av_samples_set_silence(pad_frame->extended_data, > frame->nb_samples, > + s->nb_out_samples - frame->nb_samples, > frame->channels, > + frame->format); > + av_frame_free(&frame); > + frame = pad_frame; > + } > > - ret = ff_request_frame(inlink); > - if (ret == AVERROR_EOF) { > - ret = push_samples(outlink); > - return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF; > + if (ret > 0) > + ret = ff_filter_frame(outlink, frame); > + if (ret < 0) > + return ret; Same as above: if (ret < 0) return ret; return ff_filter_frame(...); > } > > - return ret; > + FF_FILTER_FORWARD_STATUS(inlink, outlink); This should not be reached if a frame was just filtered. > + FF_FILTER_FORWARD_WANTED(outlink, inlink); > + > + return FFERROR_NOT_READY; > } > > static const AVFilterPad asetnsamples_inputs[] = { > { > - .name = "default", > - .type = AVMEDIA_TYPE_AUDIO, > - .filter_frame = filter_frame, > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > }, > { NULL } > }; > > static const AVFilterPad asetnsamples_outputs[] = { > { > - .name = "default", > - .type = AVMEDIA_TYPE_AUDIO, > - .request_frame = request_frame, > - .config_props = config_props_output, > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > }, > { NULL } > }; > @@ -191,8 +120,7 @@ AVFilter ff_af_asetnsamples = { > .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each > output audio frames."), > .priv_size = sizeof(ASNSContext), > .priv_class = &asetnsamples_class, > - .init = init, > - .uninit = uninit, > .inputs = asetnsamples_inputs, > .outputs = asetnsamples_outputs, > + .activate = activate, > }; Regards, -- Nicolas George
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