Autres/audacity
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Reimplement the insertion of zeroes at drop-outs...

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... and correct for latency only when there is at least one playback track
This commit is contained in:
Paul Licameli
2018-01-17 11:18:00 -05:00
parent be0e6cd605
commit 81b22a2b2b
3 changed files with 19 additions and 23 deletions
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38
src/AudioIO.cpp
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@@ -2716,10 +2716,6 @@ void AudioIO::StopStream()
double recordingOffset =
mLastRecordingOffset + latencyCorrection / 1000.0;
for (auto &interval : mLostCaptureIntervals)
interval.first += recordingOffset,
interval.second += recordingOffset;
for (unsigned int i = 0; i < mCaptureTracks.size(); i++) {
// The calls to Flush, and (less likely) Clear and InsertSilence,
// may cause exceptions because of exhaustion of disk space.
@@ -2791,6 +2787,19 @@ void AudioIO::StopStream()
}
} );
}
for (auto &interval : mLostCaptureIntervals) {
auto &start = interval.first;
if (mPlaybackTracks.size() > 0)
// only do latency correction if some tracks are being played back
start += recordingOffset;
auto duration = interval.second;
for (auto &track : mCaptureTracks) {
GuardedCall([&] {
track->SyncLockAdjust(start, start + duration);
});
}
}
}
}
@@ -5181,10 +5190,9 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
len < framesPerBuffer) ) {
// Assume that any good partial buffer should be written leftmost
// and zeroes will be padded after; label the zeroes.
auto start = gAudioIO->mTime;
auto end = start + framesPerBuffer / gAudioIO->mRate;
auto middle = start + len / gAudioIO->mRate;
auto interval = std::make_pair( middle, end );
auto start = gAudioIO->mTime + len / gAudioIO->mRate;
auto duration = (framesPerBuffer - len) / gAudioIO->mRate;
auto interval = std::make_pair( start, duration );
gAudioIO->mLostCaptureIntervals.push_back( interval );
}
@@ -5239,20 +5247,6 @@ int audacityAudioCallback(const void *inputBuffer, void *outputBuffer,
wxUnusedVar(put);
}
}
if (gAudioIO->mDetectDropouts &&
len < framesPerBuffer) {
for(unsigned t = 0; t < numCaptureChannels; t++) {
// Get here probably because of failure to keep up with real
// time, causing loss of input samples.
// Pad with zeroes
const auto put = gAudioIO->mCaptureBuffers[t]->Clear(
gAudioIO->mCaptureFormat, framesPerBuffer - len);
// wxASSERT(put == len);
// but we can't assert in this thread
wxUnusedVar(put);
}
}
}
// Update the current time position if not scrubbing