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https://github.com/librempeg/librempeg
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avfilter/af_dynaudnorm: allocate new frame instead of making it writable
Later case does not use frame pool at all.
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a9124a75b0
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@ -647,23 +647,42 @@ static void perform_compression(DynamicAudioNormalizerContext *s, AVFrame *frame
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}
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}
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static int analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
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static int analyze_frame(DynamicAudioNormalizerContext *s, AVFilterLink *outlink, AVFrame **frame)
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{
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if (s->dc_correction || s->compress_factor > DBL_EPSILON) {
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int ret;
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if ((ret = av_frame_make_writable(frame)) < 0)
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if (!av_frame_is_writable(*frame)) {
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AVFrame *out = ff_get_audio_buffer(outlink, (*frame)->nb_samples);
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if (!out) {
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av_frame_free(frame);
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return AVERROR(ENOMEM);
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}
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ret = av_frame_copy_props(out, *frame);
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if (ret < 0) {
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av_frame_free(&out);
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return ret;
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}
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ret = av_frame_copy(out, *frame);
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if (ret < 0) {
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av_frame_free(&out);
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return ret;
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}
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av_frame_free(frame);
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*frame = out;
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}
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}
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if (s->dc_correction)
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perform_dc_correction(s, frame);
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perform_dc_correction(s, *frame);
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if (s->compress_factor > DBL_EPSILON)
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perform_compression(s, frame);
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perform_compression(s, *frame);
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if (s->channels_coupled) {
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const local_gain gain = get_max_local_gain(s, frame, -1);
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const local_gain gain = get_max_local_gain(s, *frame, -1);
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int c;
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for (c = 0; c < s->channels; c++)
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@ -672,7 +691,7 @@ static int analyze_frame(DynamicAudioNormalizerContext *s, AVFrame *frame)
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int c;
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for (c = 0; c < s->channels; c++)
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update_gain_history(s, c, get_max_local_gain(s, frame, c));
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update_gain_history(s, c, get_max_local_gain(s, *frame, c));
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}
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return 0;
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@ -740,7 +759,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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return ret;
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}
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ret = analyze_frame(s, in);
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ret = analyze_frame(s, outlink, &in);
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if (ret < 0)
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return ret;
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if (!s->eof) {
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