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Hello, Texas Instruments,
Could you please help me to figure out one issue regarding audio codecs in DM365.
In AAC Codec documentation (/dvsdk/codecs/packages/ittiam/codecs/aac_dec/docs) it is said that "The codec handles audio signals sampled in the range of 8 kHz to 96 kHz". Does it mean that I can decode any stream with sample rate from this range and change the rate of decoded stream (within the same range)? And the same question about other audio and speech codecs?
Or, perhaps, there are any other documentation of codecs available, more detailed? Or maybe there is an xDAIS-compatible resampling algorithm?
