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https://github.com/gnif/LookingGlass.git
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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.
91 lines
2.7 KiB
C
91 lines
2.7 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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#ifndef _H_I_AUDIODEV_
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#define _H_I_AUDIODEV_
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#include <stdbool.h>
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#include <stdint.h>
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#include <stddef.h>
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typedef int (*LG_AudioPullFn)(uint8_t * dst, int frames);
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typedef void (*LG_AudioPushFn)(uint8_t * src, int frames);
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struct LG_AudioDevOps
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{
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/* internal name of the audio for debugging */
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const char * name;
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/* called very early to allow for option registration, optional */
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void (*earlyInit)(void);
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/* called to initialize the audio backend */
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bool (*init)(void);
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/* final free */
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void (*free)(void);
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struct
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{
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/* setup the stream for playback but don't start it yet, returning false
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* if the stream could not be configured
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* Note: the pull function returns f32 samples
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*/
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bool (*setup)(int channels, int sampleRate, int requestedPeriodFrames,
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int * maxPeriodFrames, int * startFrames, LG_AudioPullFn pullFn);
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/* called when there is data available to start playback */
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void (*start)(void);
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/* called when SPICE reports the audio stream has stopped */
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void (*stop)(void);
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/* [optional] called to set the volume of the channels */
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void (*volume)(int channels, const uint16_t volume[]);
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/* [optional] called to set muting of the output */
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void (*mute)(bool mute);
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/* return the current total playback latency in microseconds */
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uint64_t (*latency)(void);
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}
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playback;
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struct
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{
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/* start the record stream
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* Note: currently SPICE only supports S16 samples so always assume so
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*/
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void (*start)(int channels, int sampleRate, LG_AudioPushFn pushFn);
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/* called when SPICE reports the audio stream has stopped */
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void (*stop)(void);
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/* [optional] called to set the volume of the channels */
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void (*volume)(int channels, const uint16_t volume[]);
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/* [optional] called to set muting of the input */
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void (*mute)(bool mute);
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}
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record;
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};
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#endif
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