Paul B Mahol (2018-10-03): > Signed-off-by: Paul B Mahol <[email protected]> > --- > 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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