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@@ -96,7 +96,7 @@ static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t) |
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/* |
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* There's a couple of assumptions being made here: |
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* 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples. |
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* We can pass them upwards when ffmpeg will be ready to deal with them. |
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* We can pass them upwards when libavcodec will be ready to deal with them. |
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* 2. We don't do software emphasis. |
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* 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples |
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* are converted into 16bit linear ones. |
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