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d956dd8fd3
Drive playback timing from audio sample positions instead of packet arrival cadence. Use SPICE multimedia timestamps to anchor streams, detect discontinuities, and estimate the long-term source rate. Ignore their coarse phase corrections when steering the playback buffer. Measure the device clock independently and wait for it to stabilize before enabling a bounded, slew-limited phase controller. Align startup buffering from logical producer and consumer positions so backend startup reserves do not become persistent latency. Size the buffer from the backend period, measured delivery jitter, and resampler delay. Update PipeWire and PulseAudio to report their actual startup requirements and presentation latency. Default to a 10 ms device period with 4 ms of additional buffering, and make detailed synchronization diagnostics opt-in through audio:debug. This reduces startup and steady-state latency while compensating for physical and virtual device clock drift without reacting to transport jitter or SPICE timestamp quantization.
50 lines
1.6 KiB
C
50 lines
1.6 KiB
C
/**
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* Looking Glass
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* Copyright © 2017-2026 The Looking Glass Authors
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* https://looking-glass.io
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the Free
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* Software Foundation; either version 2 of the License, or (at your option)
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* any later version.
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*
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* This program is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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* FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
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* more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc., 59
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* Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#if ENABLE_AUDIO
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#include <stdbool.h>
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#include <purespice.h>
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void audio_init(void);
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void audio_free(void);
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bool audio_supportsPlayback(void);
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void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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uint32_t time);
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void audio_playbackStop(void);
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void audio_playbackVolume(int channels, const uint16_t volume[]);
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void audio_playbackMute(bool mute);
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void audio_playbackData(uint8_t * data, size_t size, uint32_t time);
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bool audio_supportsRecord(void);
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void audio_recordStart(int channels, int sampleRate, PSAudioFormat format);
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void audio_recordToggleKeybind(int sc, void * opaque);
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void audio_recordStop(void);
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void audio_recordVolume(int channels, const uint16_t volume[]);
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void audio_recordMute(bool mute);
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#else
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static inline void audio_init(void) {}
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static inline void audio_free(void) {}
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#endif
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