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https://github.com/xenia-project/FFmpeg.git
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avfilter/af_asetnsamples: use lavfi internal queue
This commit is contained in:
Paul B Mahol
parent
7d65fe87e9
commit
ef3babb2c7
@ -24,20 +24,18 @@
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* Filter that changes number of samples on single output operation
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "filters.h"
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#include "internal.h"
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#include "formats.h"
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typedef struct ASNSContext {
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const AVClass *class;
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int nb_out_samples; ///< how many samples to output
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AVAudioFifo *fifo; ///< samples are queued here
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int64_t next_out_pts;
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int pad;
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} ASNSContext;
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@ -54,134 +52,55 @@ static const AVOption asetnsamples_options[] = {
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AVFILTER_DEFINE_CLASS(asetnsamples);
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static av_cold int init(AVFilterContext *ctx)
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static int activate(AVFilterContext *ctx)
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{
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ASNSContext *asns = ctx->priv;
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asns->next_out_pts = AV_NOPTS_VALUE;
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av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ASNSContext *asns = ctx->priv;
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av_audio_fifo_free(asns->fifo);
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}
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static int config_props_output(AVFilterLink *outlink)
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{
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ASNSContext *asns = outlink->src->priv;
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asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, asns->nb_out_samples);
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if (!asns->fifo)
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return AVERROR(ENOMEM);
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return 0;
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}
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static int push_samples(AVFilterLink *outlink)
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{
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ASNSContext *asns = outlink->src->priv;
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AVFrame *outsamples = NULL;
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int ret, nb_out_samples, nb_pad_samples;
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if (asns->pad) {
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nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
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nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
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} else {
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nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
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nb_pad_samples = 0;
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}
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if (!nb_out_samples)
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return 0;
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outsamples = ff_get_audio_buffer(outlink, nb_out_samples);
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if (!outsamples)
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return AVERROR(ENOMEM);
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av_audio_fifo_read(asns->fifo,
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(void **)outsamples->extended_data, nb_out_samples);
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if (nb_pad_samples)
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av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
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nb_pad_samples, outlink->channels,
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outlink->format);
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outsamples->nb_samples = nb_out_samples;
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outsamples->channel_layout = outlink->channel_layout;
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outsamples->sample_rate = outlink->sample_rate;
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outsamples->pts = asns->next_out_pts;
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if (asns->next_out_pts != AV_NOPTS_VALUE)
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asns->next_out_pts += av_rescale_q(nb_out_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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ret = ff_filter_frame(outlink, outsamples);
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if (ret < 0)
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return ret;
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return nb_out_samples;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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ASNSContext *asns = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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ASNSContext *s = ctx->priv;
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AVFrame *frame = NULL, *pad_frame;
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int ret;
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int nb_samples = insamples->nb_samples;
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if (av_audio_fifo_space(asns->fifo) < nb_samples) {
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av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
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ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
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if (ret < 0) {
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av_log(ctx, AV_LOG_ERROR,
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"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
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return -1;
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}
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}
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ret = av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
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if (ret > 0 && asns->next_out_pts == AV_NOPTS_VALUE)
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asns->next_out_pts = insamples->pts;
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av_frame_free(&insamples);
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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ret = ff_inlink_consume_samples(inlink, s->nb_out_samples, s->nb_out_samples, &frame);
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if (ret < 0)
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return ret;
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while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
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push_samples(outlink);
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return 0;
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}
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if (ret > 0) {
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if ((!s->pad || (s->pad && frame->nb_samples == s->nb_out_samples)))
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return ff_filter_frame(outlink, frame);
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterLink *inlink = outlink->src->inputs[0];
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int ret;
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pad_frame = ff_get_audio_buffer(outlink, s->nb_out_samples);
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if (!pad_frame)
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return AVERROR(ENOMEM);
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ret = ff_request_frame(inlink);
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if (ret == AVERROR_EOF) {
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ret = push_samples(outlink);
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return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF;
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av_samples_copy(pad_frame->extended_data, frame->extended_data,
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0, 0, frame->nb_samples, frame->channels, frame->format);
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av_samples_set_silence(pad_frame->extended_data, frame->nb_samples,
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s->nb_out_samples - frame->nb_samples, frame->channels,
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frame->format);
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av_frame_free(&frame);
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return ff_filter_frame(outlink, pad_frame);
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}
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return ret;
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FF_FILTER_FORWARD_STATUS(inlink, outlink);
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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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static const AVFilterPad asetnsamples_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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static const AVFilterPad asetnsamples_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = request_frame,
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.config_props = config_props_output,
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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@ -191,8 +110,7 @@ AVFilter ff_af_asetnsamples = {
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.description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
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.priv_size = sizeof(ASNSContext),
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.priv_class = &asetnsamples_class,
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.init = init,
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.uninit = uninit,
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.inputs = asetnsamples_inputs,
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.outputs = asetnsamples_outputs,
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.activate = activate,
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};
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