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interpolation.h / audio_diskstream.cc: make varispeed sound well again, by replacing the code by the original implementation for later comparison and step-by-step refactoring
git-svn-id: svn://localhost/ardour2/branches/3.0@5260 d708f5d6-7413-0410-9779-e7cbd77b26cf
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417309d6d4
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2 changed files with 89 additions and 21 deletions
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@ -40,25 +40,31 @@ class FixedPointLinearInterpolation : public Interpolation {
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std::vector<uint64_t> last_phase;
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// Fixed point is just an integer with an implied scaling factor.
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// In 40.24 the scaling factor is 2^24 = 16777216,
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// so a value of 10*2^24 (in integer space) is equivalent to 10.0.
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//
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// The advantage is that addition and modulus [like x = (x + y) % 2^40]
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// have no rounding errors and no drift, and just require a single integer add.
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// (swh)
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static const int64_t fractional_part_mask = 0xFFFFFF;
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static const Sample binary_scaling_factor = 16777216.0f;
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// Fixed point is just an integer with an implied scaling factor.
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// In 40.24 the scaling factor is 2^24 = 16777216,
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// so a value of 10*2^24 (in integer space) is equivalent to 10.0.
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//
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// The advantage is that addition and modulus [like x = (x + y) % 2^40]
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// have no rounding errors and no drift, and just require a single integer add.
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// (swh)
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static const int64_t fractional_part_mask = 0xFFFFFF;
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static const Sample binary_scaling_factor = 16777216.0f;
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public:
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FixedPointLinearInterpolation () : phi (FIXPOINT_ONE), target_phi (FIXPOINT_ONE) {}
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FixedPointLinearInterpolation () : phi (FIXPOINT_ONE), target_phi (FIXPOINT_ONE) {}
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void set_speed (double new_speed) {
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target_phi = (uint64_t) (FIXPOINT_ONE * fabs(new_speed));
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phi = target_phi;
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}
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uint64_t get_phi() { return phi; }
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uint64_t get_target_phi() { return target_phi; }
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uint64_t get_last_phase() { assert(last_phase.size()); return last_phase[0]; }
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void set_last_phase(uint64_t phase) { assert(last_phase.size()); last_phase[0] = phase; }
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void add_channel_to (int input_buffer_size, int output_buffer_size);
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void remove_channel_from ();
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@ -816,18 +816,80 @@ AudioDiskstream::process (nframes_t transport_frame, nframes_t nframes, bool can
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void
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AudioDiskstream::process_varispeed_playback(nframes_t nframes, boost::shared_ptr<ChannelList> c)
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{
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ChannelList::iterator chan;
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interpolation.set_target_speed (_target_speed);
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interpolation.set_speed (_speed);
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ChannelList::iterator chan;
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int channel = 0;
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for (chan = c->begin(); chan != c->end(); ++chan, ++channel) {
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ChannelInfo* chaninfo (*chan);
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/*
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interpolation.set_speed (_target_speed);
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int channel = 0;
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for (chan = c->begin(); chan != c->end(); ++chan, ++channel) {
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ChannelInfo* chaninfo (*chan);
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playback_distance = interpolation.interpolate (
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channel, nframes, chaninfo->current_playback_buffer, chaninfo->speed_buffer);
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}
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*/
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// the idea behind phase is that when the speed is not 1.0, we have to
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// interpolate between samples and then we have to store where we thought we were.
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// rather than being at sample N or N+1, we were at N+0.8792922
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// so the "phase" element, if you want to think about this way,
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// varies from 0 to 1, representing the "offset" between samples
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uint64_t phase = interpolation.get_last_phase();
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interpolation.set_speed (_target_speed);
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// acceleration
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uint64_t phi = interpolation.get_phi();
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uint64_t target_phi = interpolation.get_target_phi();
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int64_t phi_delta;
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// index in the input buffers
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nframes_t i = 0;
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playback_distance = interpolation.interpolate (
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channel, nframes, chaninfo->current_playback_buffer, chaninfo->speed_buffer);
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}
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// Linearly interpolate into the speed buffer
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// using 40.24 fixed point math
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//
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// Fixed point is just an integer with an implied scaling factor.
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// In 40.24 the scaling factor is 2^24 = 16777216,
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// so a value of 10*2^24 (in integer space) is equivalent to 10.0.
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//
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// The advantage is that addition and modulus [like x = (x + y) % 2^40]
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// have no rounding errors and no drift, and just require a single integer add.
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// (swh)
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const int64_t fractional_part_mask = 0xFFFFFF;
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const Sample binary_scaling_factor = 16777216.0f;
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// phi = fixed point speed
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if (phi != target_phi) {
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phi_delta = ((int64_t)(target_phi - phi)) / nframes;
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} else {
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phi_delta = 0;
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}
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for (chan = c->begin(); chan != c->end(); ++chan) {
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Sample fractional_phase_part;
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ChannelInfo* chaninfo (*chan);
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i = 0;
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phase = interpolation.get_last_phase();
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for (nframes_t outsample = 0; outsample < nframes; ++outsample) {
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i = phase >> 24;
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fractional_phase_part = (phase & fractional_part_mask) / binary_scaling_factor;
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chaninfo->speed_buffer[outsample] =
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chaninfo->current_playback_buffer[i] * (1.0f - fractional_phase_part) +
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chaninfo->current_playback_buffer[i+1] * fractional_phase_part;
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phase += phi + phi_delta;
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}
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chaninfo->current_playback_buffer = chaninfo->speed_buffer;
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}
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playback_distance = i; // + 1;
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interpolation.set_last_phase (phase & fractional_part_mask);
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}
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bool
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