X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavfilter%2Faf_amerge.c;h=567f25982d8ff028f1e299414446051f773d8038;hb=7d5bb3a4d34f0cddd300252f8afc37c3d7bf17c8;hp=09c660ef49b5566aa24088e6684fb8f8b6aabdeb;hpb=52b44e9d15c0ee3c118ed68a0c2c737a9eb50ae9;p=ffmpeg diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c index 09c660ef49b..567f25982d8 100644 --- a/libavfilter/af_amerge.c +++ b/libavfilter/af_amerge.c @@ -23,16 +23,13 @@ * Audio merging filter */ -#define FF_INTERNAL_FIELDS 1 -#include "framequeue.h" - #include "libavutil/avstring.h" #include "libavutil/bprint.h" #include "libavutil/channel_layout.h" #include "libavutil/opt.h" #include "avfilter.h" +#include "filters.h" #include "audio.h" -#include "bufferqueue.h" #include "internal.h" #define SWR_CH_MAX 64 @@ -43,10 +40,7 @@ typedef struct AMergeContext { int route[SWR_CH_MAX]; /**< channels routing, see copy_samples */ int bps; struct amerge_input { - struct FFBufQueue queue; int nb_ch; /**< number of channels for the input */ - int nb_samples; - int pos; } *in; } AMergeContext; @@ -67,8 +61,6 @@ static av_cold void uninit(AVFilterContext *ctx) int i; for (i = 0; i < s->nb_inputs; i++) { - if (s->in) - ff_bufqueue_discard_all(&s->in[i].queue); if (ctx->input_pads) av_freep(&ctx->input_pads[i].name); } @@ -171,7 +163,7 @@ static int config_output(AVFilterLink *outlink) outlink->sample_rate = ctx->inputs[0]->sample_rate; outlink->time_base = ctx->inputs[0]->time_base; - av_bprint_init(&bp, 0, 1); + av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC); for (i = 0; i < s->nb_inputs; i++) { av_bprintf(&bp, "%sin%d:", i ? " + " : "", i); av_bprint_channel_layout(&bp, -1, ctx->inputs[i]->channel_layout); @@ -183,21 +175,6 @@ static int config_output(AVFilterLink *outlink) return 0; } -static int request_frame(AVFilterLink *outlink) -{ - AVFilterContext *ctx = outlink->src; - AMergeContext *s = ctx->priv; - int i, ret; - - for (i = 0; i < s->nb_inputs; i++) - if (!s->in[i].nb_samples || - /* detect EOF immediately */ - (ctx->inputs[i]->status_in && !ctx->inputs[i]->status_out)) - if ((ret = ff_request_frame(ctx->inputs[i])) < 0) - return ret; - return 0; -} - /** * Copy samples from several input streams to one output stream. * @param nb_inputs number of inputs @@ -235,90 +212,103 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[], } } -static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) +static void free_frames(int nb_inputs, AVFrame **input_frames) +{ + int i; + for (i = 0; i < nb_inputs; i++) + av_frame_free(&input_frames[i]); +} + +static int try_push_frame(AVFilterContext *ctx, int nb_samples) { - AVFilterContext *ctx = inlink->dst; AMergeContext *s = ctx->priv; - AVFilterLink *const outlink = ctx->outputs[0]; - int input_number; - int nb_samples, ns, i; - AVFrame *outbuf, *inbuf[SWR_CH_MAX]; - uint8_t *ins[SWR_CH_MAX], *outs; - - for (input_number = 0; input_number < s->nb_inputs; input_number++) - if (inlink == ctx->inputs[input_number]) - break; - av_assert1(input_number < s->nb_inputs); - if (ff_bufqueue_is_full(&s->in[input_number].queue)) { - av_frame_free(&insamples); - return AVERROR(ENOMEM); + AVFilterLink *outlink = ctx->outputs[0]; + int i, ret; + AVFrame *outbuf, *inbuf[SWR_CH_MAX] = { NULL }; + uint8_t *outs, *ins[SWR_CH_MAX]; + + for (i = 0; i < ctx->nb_inputs; i++) { + ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &inbuf[i]); + if (ret < 0) { + free_frames(i, inbuf); + return ret; + } + ins[i] = inbuf[i]->data[0]; } - ff_bufqueue_add(ctx, &s->in[input_number].queue, av_frame_clone(insamples)); - s->in[input_number].nb_samples += insamples->nb_samples; - av_frame_free(&insamples); - nb_samples = s->in[0].nb_samples; - for (i = 1; i < s->nb_inputs; i++) - nb_samples = FFMIN(nb_samples, s->in[i].nb_samples); - if (!nb_samples) - return 0; outbuf = ff_get_audio_buffer(ctx->outputs[0], nb_samples); - if (!outbuf) + if (!outbuf) { + free_frames(s->nb_inputs, inbuf); return AVERROR(ENOMEM); - outs = outbuf->data[0]; - for (i = 0; i < s->nb_inputs; i++) { - inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0); - ins[i] = inbuf[i]->data[0] + - s->in[i].pos * s->in[i].nb_ch * s->bps; } - av_frame_copy_props(outbuf, inbuf[0]); - outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE : - inbuf[0]->pts + - av_rescale_q(s->in[0].pos, - av_make_q(1, ctx->inputs[0]->sample_rate), - ctx->outputs[0]->time_base); + + outs = outbuf->data[0]; + outbuf->pts = inbuf[0]->pts; outbuf->nb_samples = nb_samples; outbuf->channel_layout = outlink->channel_layout; outbuf->channels = outlink->channels; while (nb_samples) { - ns = nb_samples; - for (i = 0; i < s->nb_inputs; i++) - ns = FFMIN(ns, inbuf[i]->nb_samples - s->in[i].pos); /* Unroll the most common sample formats: speed +~350% for the loop, +~13% overall (including two common decoders) */ switch (s->bps) { case 1: - copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 1); + copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 1); break; case 2: - copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 2); + copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 2); break; case 4: - copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, 4); + copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, 4); break; default: - copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, ns, s->bps); + copy_samples(s->nb_inputs, s->in, s->route, ins, &outs, nb_samples, s->bps); break; } - nb_samples -= ns; - for (i = 0; i < s->nb_inputs; i++) { - s->in[i].nb_samples -= ns; - s->in[i].pos += ns; - if (s->in[i].pos == inbuf[i]->nb_samples) { - s->in[i].pos = 0; - av_frame_free(&inbuf[i]); - ff_bufqueue_get(&s->in[i].queue); - inbuf[i] = ff_bufqueue_peek(&s->in[i].queue, 0); - ins[i] = inbuf[i] ? inbuf[i]->data[0] : NULL; - } - } + nb_samples = 0; } + + free_frames(s->nb_inputs, inbuf); return ff_filter_frame(ctx->outputs[0], outbuf); } +static int activate(AVFilterContext *ctx) +{ + int i, status; + int ret, nb_samples; + int64_t pts; + + FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); + + nb_samples = ff_inlink_queued_samples(ctx->inputs[0]); + for (i = 1; i < ctx->nb_inputs && nb_samples > 0; i++) { + nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[i]), nb_samples); + } + + if (nb_samples) { + ret = try_push_frame(ctx, nb_samples); + if (ret < 0) + return ret; + } + + for (i = 0; i < ctx->nb_inputs; i++) { + if (ff_inlink_queued_samples(ctx->inputs[i])) + continue; + + if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { + ff_outlink_set_status(ctx->outputs[0], status, pts); + return 0; + } else if (ff_outlink_frame_wanted(ctx->outputs[0])) { + ff_inlink_request_frame(ctx->inputs[i]); + return 0; + } + } + + return 0; +} + static av_cold int init(AVFilterContext *ctx) { AMergeContext *s = ctx->priv; @@ -332,7 +322,6 @@ static av_cold int init(AVFilterContext *ctx) AVFilterPad pad = { .name = name, .type = AVMEDIA_TYPE_AUDIO, - .filter_frame = filter_frame, }; if (!name) return AVERROR(ENOMEM); @@ -349,7 +338,6 @@ static const AVFilterPad amerge_outputs[] = { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, - .request_frame = request_frame, }, { NULL } }; @@ -362,6 +350,7 @@ AVFilter ff_af_amerge = { .init = init, .uninit = uninit, .query_formats = query_formats, + .activate = activate, .inputs = NULL, .outputs = amerge_outputs, .priv_class = &amerge_class,