Mike-Solar
a7ddc0f114
- playback timer uses the audio output device as its master clock: the PortAudio callback counts consumed frames (including underrun zero-fill) so video cannot drift away from what is heard; wall clock remains as fallback when no clocked output is running - output clock compensates for device output latency; new Preferences > Audio buffer size setting (0 = auto) - SampleBuffer::speed() now uses linear interpolation instead of nearest-neighbor sampling - regression tests: audio-clock driven timer (fwd/rev/speed), wall clock fallback, interpolation correctness
70 lines
2.2 KiB
C++
70 lines
2.2 KiB
C++
/***
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Olive - Non-Linear Video Editor
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Copyright (C) 2022 Olive Team
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Modifications Copyright (C) 2025 mikesolar
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program. If not, see <http://www.gnu.org/licenses/>.
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***/
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#include "viewerplaybacktimer.h"
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#include <QtMath>
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namespace olive
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{
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void ViewerPlaybackTimer::start(const int64_t &start_timestamp,
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const int &playback_speed,
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const double &timebase,
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const PlaybackAudioClock *audio_clock)
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{
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timer_.start();
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start_timestamp_ = start_timestamp;
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playback_speed_ = playback_speed;
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timebase_ = timebase * 1000;
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audio_clock_ = audio_clock;
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}
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int64_t ViewerPlaybackTimer::get_timestamp_now() const
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{
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// The audio output clock is the master clock when available: sound card
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// consumption is what the viewer must stay in sync with, and unlike the
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// wall clock it cannot drift away from what is actually heard
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if (audio_clock_) {
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const double audio_seconds = audio_clock_->seconds();
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if (audio_seconds >= 0.0) {
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// At speeds other than 1x the audio tempo is scaled, so one
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// output second corresponds to |speed| timeline seconds. The
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// result is already in timeline frames, so it is applied
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// signed rather than multiplied by the speed again.
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const double timeline_ms =
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audio_seconds * 1000.0 * qAbs(playback_speed_);
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const int64_t frames_since_start = qFloor(timeline_ms / timebase_);
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return start_timestamp_ +
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frames_since_start * (playback_speed_ < 0 ? -1 : 1);
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}
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}
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int64_t real_time = timer_.elapsed();
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int64_t frames_since_start =
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qFloor(static_cast<double>(real_time) / (timebase_));
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return start_timestamp_ + frames_since_start * playback_speed_;
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}
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}
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