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@@ -67,7 +67,7 @@ void AudioTuner::update( void ) { |
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release( block ); |
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if ( count >= NUM_SAMPLES ) { |
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digitalWriteFast(2, !digitalReadFast(2)); |
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//digitalWriteFast(2, !digitalReadFast(2)); |
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next_buffer = !next_buffer; |
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process_buffer = true; |
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tau_global = 1; |
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@@ -78,7 +78,7 @@ void AudioTuner::update( void ) { |
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block_count = count; |
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if ( process_buffer ) { |
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digitalWriteFast(0, HIGH); |
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//digitalWriteFast(0, HIGH); |
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uint16_t tau; |
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uint16_t next; |
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next = next_buffer; |
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@@ -107,23 +107,30 @@ void AudioTuner::update( void ) { |
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if ( tau == 0 ) { |
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process_buffer = false; |
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new_output = true; |
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digitalWriteFast(0, LOW); |
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//digitalWriteFast(0, LOW); |
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return; |
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} |
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else if ( tau >= HALF_BUFFER ) { |
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process_buffer = false; |
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new_output = false; |
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digitalWriteFast(0, LOW); |
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//digitalWriteFast(0, LOW); |
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return; |
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} |
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} while ( tau <= ( tau_global + 31 ) ); |
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tau_global = tau; |
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digitalWriteFast(0, LOW); |
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//digitalWriteFast(0, LOW); |
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} |
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} |
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/** |
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* process data in from Audio Library interrupt |
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* check the sampled data for fundmental frequency |
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* |
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* @param yin buffer to hold sum*tau value |
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* @param rs buffer to hold running sum for sampled window |
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* @param head buffer index |
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* @param tau lag we are currently working on this gets incremented |
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* |
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* @return tau |
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*/ |
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uint16_t AudioTuner::estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau ) { |
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const int64_t *p = ( int64_t * )yin; |