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

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from AudioDiskstream::process


git-svn-id: svn://localhost/ardour2/branches/3.0@4396 d708f5d6-7413-0410-9779-e7cbd77b26cf
This commit is contained in:
Hans Baier 2009-01-10 08:41:51 +00:00
parent 3d2c1ba3e6
commit bfbae251be
2 changed files with 68 additions and 55 deletions
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5
libs/ardour/ardour/audio_diskstream.h
View File

@ -175,7 +175,7 @@ class AudioDiskstream : public Diskstream
protected:
friend class AudioTrack;
int process (nframes_t transport_frame, nframes_t nframes, nframes_t offset, bool can_record, bool rec_monitors_input);
int process (nframes_t transport_frame, nframes_t nframes, nframes_t offset, bool can_record, bool rec_monitors_input);
bool commit (nframes_t nframes);
private:
@ -216,6 +216,8 @@ class AudioDiskstream : public Diskstream
typedef std::vector<ChannelInfo*> ChannelList;
void process_varispeed_playback(nframes_t nframes, boost::shared_ptr<ChannelList> c);
/* The two central butler operations */
int do_flush (Session::RunContext context, bool force = false);
int do_refill () { return _do_refill(_mixdown_buffer, _gain_buffer); }
@ -226,6 +228,7 @@ class AudioDiskstream : public Diskstream
nframes_t& start, nframes_t cnt,
ChannelInfo* channel_info, int channel, bool reversed);
void finish_capture (bool rec_monitors_input, boost::shared_ptr<ChannelList>);
void transport_stopped (struct tm&, time_t, bool abort);
void transport_looped (nframes_t transport_frame);

118
libs/ardour/audio_diskstream.cc
View File

@ -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)
{