* This class takes an input audio signal and a sequence of beat
* locations (calculated e.g. by TempoTrackV2) and estimates which of
* the beat locations are downbeats (first beat of the bar).
- *
+ *
* The input audio signal is expected to have been downsampled to a
* very low sampling rate (e.g. 2700Hz). A utility function for
* downsampling and buffering incoming block-by-block audio is
/**
* Estimate which beats are down-beats.
- *
+ *
* audio contains the input audio stream after downsampling, and
* audioLength contains the number of samples in this downsampled
* stream.
* and the region following it.
*/
void getBeatSD(vector<double> &beatsd) const;
-
+
/**
* For your downsampling convenience: call this function
* repeatedly with input audio blocks containing dfIncrement
* samples at the original sample rate, to decimate them to the
* downsampled rate and buffer them within the DownBeat class.
- *
+ *
* Call getBufferedAudio() to retrieve the results after all
* blocks have been processed.
*/
void pushAudioBlock(const float *audio);
-
+
/**
* Retrieve the accumulated audio produced by pushAudioBlock calls.
*/