mirror of
https://github.com/gnif/LookingGlass.git
synced 2024-11-22 05:27:20 +00:00
[client] audio: change the audio latency graph sample point
This removes the need for locking while also giving a better result in the graph output. Also when the graph is disabled via the overlay options it will no longer cause redraws.
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a0477466d2
commit
22b968ff53
@ -116,7 +116,7 @@ typedef const char * (*GraphFormatFn)(const char * name,
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GraphHandle app_registerGraph(const char * name, RingBuffer buffer,
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float min, float max, GraphFormatFn formatFn);
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void app_unregisterGraph(GraphHandle handle);
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void app_invalidateGraphs(void);
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void app_invalidateGraph(GraphHandle handle);
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void app_overlayConfigRegister(const char * title,
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void (*callback)(void * udata, int * id), void * udata);
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@ -706,9 +706,9 @@ void app_unregisterGraph(GraphHandle handle)
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overlayGraph_unregister(handle);
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}
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void app_invalidateGraphs(void)
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void app_invalidateGraph(GraphHandle handle)
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{
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overlayGraph_invalidate();
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overlayGraph_invalidate(handle);
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}
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void app_registerOverlay(const struct LG_OverlayOps * ops, const void * params)
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@ -100,7 +100,6 @@ typedef struct
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RingBuffer buffer;
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RingBuffer deviceTiming;
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LG_Lock lock;
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RingBuffer timings;
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GraphHandle graph;
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@ -134,7 +133,7 @@ typedef struct
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}
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PlaybackDeviceTick;
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static void playbackStopNL(void);
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static void playbackStop(void);
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void audio_init(void)
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{
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@ -143,7 +142,6 @@ void audio_init(void)
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if (LG_AudioDevs[i]->init())
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{
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audio.audioDev = LG_AudioDevs[i];
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LG_LOCK_INIT(audio.playback.lock);
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DEBUG_INFO("Using AudioDev: %s", audio.audioDev->name);
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return;
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}
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@ -157,15 +155,11 @@ void audio_free(void)
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return;
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// immediate stop of the stream, do not wait for drain
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LG_LOCK(audio.playback.lock);
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playbackStopNL();
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LG_UNLOCK(audio.playback.lock);
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playbackStop();
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audio_recordStop();
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audio.audioDev->free();
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audio.audioDev = NULL;
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LG_LOCK_FREE(audio.playback.lock);
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}
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bool audio_supportsPlayback(void)
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@ -183,7 +177,7 @@ static const char * audioGraphFormatFn(const char * name,
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return title;
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}
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static void playbackStopNL(void)
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static void playbackStop(void)
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{
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if (audio.playback.state == STREAM_STATE_STOP)
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return;
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@ -274,11 +268,8 @@ static int playbackPullFrames(uint8_t * dst, int frames)
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if (audio.playback.state == STREAM_STATE_DRAIN &&
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ringbuffer_getCount(audio.playback.buffer) <= 0)
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{
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LG_LOCK(audio.playback.lock);
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playbackStopNL();
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LG_UNLOCK(audio.playback.lock);
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}
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playbackStop();
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return frames;
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}
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@ -288,13 +279,11 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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if (!audio.audioDev)
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return;
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LG_LOCK(audio.playback.lock);
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if (audio.playback.state != STREAM_STATE_STOP)
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{
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// Stop the current playback immediately. Even if the format is compatible,
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// we may not have enough data left in the buffers to avoid underrunning
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playbackStopNL();
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playbackStop();
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}
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int srcError;
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@ -303,7 +292,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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if (!audio.playback.spiceData.src)
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{
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DEBUG_ERROR("Failed to create resampler: %s", src_strerror(srcError));
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goto done;
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return;
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}
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const int bufferFrames = sampleRate;
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@ -346,9 +335,6 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
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audio.playback.state = STREAM_STATE_SETUP;
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done:
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LG_UNLOCK(audio.playback.lock);
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}
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void audio_playbackStop(void)
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@ -418,7 +404,7 @@ void audio_playbackData(uint8_t * data, size_t size)
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if (!spiceData->framesIn)
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{
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DEBUG_ERROR("Failed to malloc framesIn");
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playbackStopNL();
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playbackStop();
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return;
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}
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@ -428,7 +414,7 @@ void audio_playbackData(uint8_t * data, size_t size)
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if (!spiceData->framesOut)
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{
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DEBUG_ERROR("Failed to malloc framesOut");
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playbackStopNL();
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playbackStop();
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return;
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}
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}
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@ -571,6 +557,15 @@ void audio_playbackData(uint8_t * data, size_t size)
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if (audio.audioDev->playback.start(frames))
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audio.playback.state = STREAM_STATE_RUN;
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}
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int latencyFrames = ringbuffer_getCount(audio.playback.buffer);
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if (audio.audioDev->playback.latency)
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latencyFrames += audio.audioDev->playback.latency();
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const float latency = latencyFrames /
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(float)(audio.playback.sampleRate / 1000);
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ringbuffer_push(audio.playback.timings, &latency);
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app_invalidateGraph(audio.playback.graph);
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}
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bool audio_supportsRecord(void)
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@ -655,26 +650,4 @@ void audio_recordMute(bool mute)
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audio.audioDev->record.mute(mute);
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}
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void audio_tick(unsigned long long tickCount)
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{
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LG_LOCK(audio.playback.lock);
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if (!audio.playback.buffer)
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{
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LG_UNLOCK(audio.playback.lock);
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return;
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}
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int frames = ringbuffer_getCount(audio.playback.buffer);
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if (audio.audioDev->playback.latency)
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frames += audio.audioDev->playback.latency();
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const float latency = frames / (float)(audio.playback.sampleRate / 1000);
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ringbuffer_push(audio.playback.timings, &latency);
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LG_UNLOCK(audio.playback.lock);
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app_invalidateGraphs();
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}
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#endif
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@ -40,12 +40,9 @@ 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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void audio_tick(unsigned long long tickCount);
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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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static inline void audio_tick(unsigned long long tickCount) {}
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#endif
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@ -155,8 +155,6 @@ static bool tickTimerFn(void * unused)
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if (needsRender)
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app_invalidateWindow(false);
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audio_tick(tickCount);
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++tickCount;
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return true;
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}
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@ -245,10 +245,11 @@ void overlayGraph_iterate(void (*callback)(GraphHandle handle, const char * name
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ll_unlock(gs.graphs);
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}
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void overlayGraph_invalidate(void)
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void overlayGraph_invalidate(GraphHandle handle)
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{
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if (!gs.show)
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return;
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app_invalidateWindow(false);
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if (handle->enabled)
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app_invalidateWindow(false);
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}
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@ -47,7 +47,7 @@ GraphHandle overlayGraph_register(const char * name, RingBuffer buffer,
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void overlayGraph_unregister();
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void overlayGraph_iterate(void (*callback)(GraphHandle handle, const char * name,
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bool * enabled, void * udata), void * udata);
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void overlayGraph_invalidate(void);
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void overlayGraph_invalidate(GraphHandle handle);
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void overlayConfig_register(const char * title,
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void (*callback)(void * udata, int * id), void * udata);
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