* Extracted method void AudioDiskstream::process_varispeed_playback(nframes_t nframes, boost::shared_ptr<ChannelList> c)

  from AudioDiskstream::process


git-svn-id: svn://localhost/ardour2/branches/3.0@4396 d708f5d6-7413-0410-9779-e7cbd77b26cf

1 files changed, 64 insertions, 54 deletions

diff --git a/libs/ardour/audio_diskstream.cc b/libs/ardour/audio_diskstream.cc
index cd2148325a..36903f7699 100644
--- a/libs/ardour/audio_diskstream.cc
+++ b/libs/ardour/audio_diskstream.cc
@@ -779,60 +779,7 @@ AudioDiskstream::process (nframes_t transport_frame, nframes_t nframes, nframes_
 		} 
 
 		if (rec_nframes == 0 && _actual_speed != 1.0f && _actual_speed != -1.0f) {
-			
-			// the idea behind phase is that when the speed is not 1.0, we have to 
-			// interpolate between samples and then we have to store where we thought we were. 
-			// rather than being at sample N or N+1, we were at N+0.8792922
-			// so the "phase" element, if you want to think about this way, 
-			// varies from 0 to 1, representing the "offset" between samples
-			uint64_t phase = last_phase;
-			int64_t phi_delta;
-			nframes_t i = 0;
-
-			// Linearly interpolate into the alt buffer
-			// using 40.24 fixp maths
-			//
-			// Fixedpoint is just an integer with an implied scaling factor. 
-			// In 40.24 the scaling factor is 2^24 = 16777216,  
-			// so a value of 10*2^24 (in integer space) is equivalent to 10.0. 
-			//
-			// The advantage is that addition and modulus [like x = (x + y) % 2^40]  
-			// has no rounding errors and no drift, and just requires a single integer add.
-			// (swh)
-			
-			const int64_t fractional_part_mask  = 0xFFFFFF;
-			const Sample  binary_scaling_factor = 16777216.0f;
-
-			// phi = fixed point speed
-			if (phi != target_phi) {
-				phi_delta = ((int64_t)(target_phi - phi)) / nframes;
-			} else {
-				phi_delta = 0;
-			}
-
-			for (chan = c->begin(); chan != c->end(); ++chan) {
-
-				Sample fractional_part;
-				ChannelInfo* chaninfo (*chan);
-
-				i = 0;
-				phase = last_phase;
-
-				for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
-					i = phase >> 24;
-					fractional_part = (phase & fractional_part_mask) / binary_scaling_factor;
-					chaninfo->speed_buffer[outsample] = 
-						chaninfo->current_playback_buffer[i] * (1.0f - fractional_part) +
-						chaninfo->current_playback_buffer[i+1] * fractional_part;
-					phase += phi + phi_delta;
-				}
-				
-				chaninfo->current_playback_buffer = chaninfo->speed_buffer;
-			}
-
-			playback_distance = i; // + 1;
-			last_phase = (phase & fractional_part_mask);
-
+			process_varispeed_playback(nframes, c);
 		} else {
 			playback_distance = nframes;
 		}
@@ -859,6 +806,69 @@ AudioDiskstream::process (nframes_t transport_frame, nframes_t nframes, nframes_
 	return ret;
 }
 
+void
+AudioDiskstream::process_varispeed_playback(nframes_t nframes, boost::shared_ptr<ChannelList> c)
+{
+	ChannelList::iterator chan;
+	
+	// the idea behind phase is that when the speed is not 1.0, we have to 
+	// interpolate between samples and then we have to store where we thought we were. 
+	// rather than being at sample N or N+1, we were at N+0.8792922
+	// so the "phase" element, if you want to think about this way, 
+	// varies from 0 to 1, representing the "offset" between samples
+	uint64_t    phase = last_phase;
+	
+	// acceleration
+	int64_t     phi_delta;
+	
+	// index in the input buffers
+	nframes_t   i = 0;
+
+	// Linearly interpolate into the speed buffer
+	// using 40.24 fixed point math
+	//
+	// Fixed point is just an integer with an implied scaling factor. 
+	// In 40.24 the scaling factor is 2^24 = 16777216,  
+	// so a value of 10*2^24 (in integer space) is equivalent to 10.0. 
+	//
+	// The advantage is that addition and modulus [like x = (x + y) % 2^40]  
+	// have no rounding errors and no drift, and just require a single integer add.
+	// (swh)
+	
+	const int64_t fractional_part_mask  = 0xFFFFFF;
+	const Sample  binary_scaling_factor = 16777216.0f;
+
+	// phi = fixed point speed
+	if (phi != target_phi) {
+		phi_delta = ((int64_t)(target_phi - phi)) / nframes;
+	} else {
+		phi_delta = 0;
+	}
+
+	for (chan = c->begin(); chan != c->end(); ++chan) {
+
+		Sample fractional_phase_part;
+		ChannelInfo* chaninfo (*chan);
+
+		i = 0;
+		phase = last_phase;
+
+		for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
+			i = phase >> 24;
+			fractional_phase_part = (phase & fractional_part_mask) / binary_scaling_factor;
+			chaninfo->speed_buffer[outsample] = 
+				chaninfo->current_playback_buffer[i] * (1.0f - fractional_phase_part) +
+				chaninfo->current_playback_buffer[i+1] * fractional_phase_part;
+			phase += phi + phi_delta;
+		}
+		
+		chaninfo->current_playback_buffer = chaninfo->speed_buffer;
+	}
+
+	playback_distance = i; // + 1;
+	last_phase = (phase & fractional_part_mask);
+}
+
 bool
 AudioDiskstream::commit (nframes_t nframes)
 {