update filtering_audio example
The filtering_audio.c example needs to be updated for the new "abuffer" filter args. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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@ -3,7 +3,6 @@ FFMPEG_LIBS= libavdevice \
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libavformat \
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libavfilter \
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libavcodec \
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libavresample \
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libswresample \
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libswscale \
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libavutil \
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@ -91,14 +91,17 @@ static int init_filters(const char *filters_descr)
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const int64_t *chlayouts = avfilter_all_channel_layouts;
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AVABufferSinkParams *abuffersink_params;
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const AVFilterLink *outlink;
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AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
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filter_graph = avfilter_graph_alloc();
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/* buffer audio source: the decoded frames from the decoder will be inserted here. */
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if (!dec_ctx->channel_layout)
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dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
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snprintf(args, sizeof(args), "%d:%d:0x%"PRIx64,
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dec_ctx->sample_rate, dec_ctx->sample_fmt, dec_ctx->channel_layout);
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snprintf(args, sizeof(args),
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"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
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time_base.num, time_base.den, dec_ctx->sample_rate,
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av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
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ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
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args, NULL, filter_graph);
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if (ret < 0) {
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