[FFmpeg-devel] [PATCH 2/4] avfilter/af_asetnsamples: use lavfi internal queue
Nicolas George
george at nsup.org
Wed Oct 3 17:52:54 EEST 2018
Paul B Mahol (2018-10-03):
> Signed-off-by: Paul B Mahol <onemda at 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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