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https://github.com/xenia-project/FFmpeg.git
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apad: add whole_len and pad_len options.
These options allow padding to a specific length or adding a specific amount if silence. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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c2640146e0
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c311431e29
@ -40,6 +40,8 @@ typedef struct {
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int64_t next_pts;
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int packet_size;
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int64_t pad_len;
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int64_t whole_len;
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} APadContext;
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#define OFFSET(x) offsetof(APadContext, x)
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@ -47,6 +49,8 @@ typedef struct {
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static const AVOption apad_options[] = {
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{ "packet_size", "set silence packet size", OFFSET(packet_size), AV_OPT_TYPE_INT, { .i64 = 4096 }, 0, INT_MAX, A },
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{ "pad_len", "number of samples of silence to add", OFFSET(pad_len), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
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{ "whole_len", "target number of samples in the audio stream", OFFSET(whole_len), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT64_MAX, A },
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{ NULL },
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};
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@ -65,6 +69,11 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
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if ((ret = av_opt_set_from_string(apad, args, NULL, "=", ":")) < 0)
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return ret;
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if (apad->whole_len && apad->pad_len) {
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av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
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return AVERROR(EINVAL);
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}
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av_opt_free(apad);
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return 0;
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@ -74,6 +83,10 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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APadContext *apad = ctx->priv;
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if (apad->whole_len)
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apad->whole_len -= frame->audio->nb_samples;
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apad->next_pts = frame->pts + av_rescale_q(frame->audio->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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return ff_filter_frame(ctx->outputs[0], frame);
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}
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@ -88,7 +101,21 @@ static int request_frame(AVFilterLink *outlink)
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if (ret == AVERROR_EOF) {
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int n_out = apad->packet_size;
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AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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AVFilterBufferRef *outsamplesref;
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if (apad->whole_len > 0) {
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apad->pad_len = apad->whole_len;
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apad->whole_len = 0;
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}
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if (apad->pad_len > 0) {
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n_out = FFMIN(n_out, apad->pad_len);
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apad->pad_len -= n_out;
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}
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if(!n_out)
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return AVERROR_EOF;
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outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
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if (!outsamplesref)
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return AVERROR(ENOMEM);
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