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Revert "[client] audio: tune the target latency based on the latency jitter"
This reverts commit febd081202
.
This causes severe underruns when the quantum size increases.
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35bf30910b
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ca29fe80a6
@ -102,7 +102,6 @@ typedef struct
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RingBuffer timings;
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GraphHandle graph;
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float jitter;
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// These two structs contain data specifically for use in the device and
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// Spice data threads respectively. Keep them on separate cache lines to
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@ -173,8 +172,8 @@ static const char * audioGraphFormatFn(const char * name,
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{
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static char title[64];
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snprintf(title, sizeof(title),
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"%s: min:%4.2f max:%4.2f avg:%4.2f now:%4.2f jitter:%4.2f",
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name, min, max, avg, last, max - min);
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"%s: min:%4.2f max:%4.2f avg:%4.2f now:%4.2f",
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name, min, max, avg, last);
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return title;
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}
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@ -306,7 +305,6 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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audio.playback.sampleRate = sampleRate;
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audio.playback.stride = channels * sizeof(float);
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audio.playback.state = STREAM_STATE_SETUP;
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audio.playback.jitter = 60.0f; //assume 60ms of jitter initially
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audio.playback.deviceData.periodFrames = 0;
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audio.playback.deviceData.nextPosition = 0;
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@ -331,10 +329,8 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
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if (audio.audioDev->playback.mute)
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audio.audioDev->playback.mute(audio.playback.mute);
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// timings for jitter calculations and display graph
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// spice operates on a period size of (sampleRate / 100), so allocate enough
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// room for 4 seconds of timing samples.
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audio.playback.timings = ringbuffer_new(sampleRate / 100, sizeof(float));
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// if the audio dev can report it's latency setup a timing graph
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audio.playback.timings = ringbuffer_new(1200, sizeof(float));
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audio.playback.graph = app_registerGraph("PLAYBACK",
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audio.playback.timings, 0.0f, 100.0f, audioGraphFormatFn);
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@ -379,22 +375,6 @@ void audio_playbackMute(bool mute)
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audio.audioDev->playback.mute(mute);
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}
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static bool getMinMax(int index, void * value, void * udata)
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{
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float ms = *(float *)value;
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float * minMax = (float *)udata;
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if (index == 0)
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{
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minMax[0] = minMax[1] = ms;
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return true;
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}
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minMax[0] = min(minMax[0], ms);
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minMax[1] = max(minMax[1], ms);
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return true;
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}
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void audio_playbackData(uint8_t * data, size_t size)
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{
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if (!audio.audioDev || size == 0)
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@ -518,7 +498,7 @@ void audio_playbackData(uint8_t * data, size_t size)
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// device period size, but that would result in underruns if the period size
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// suddenly increases. It may be better instead to just reduce the maximum
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// latency on the audio devices, which currently is set quite high
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int targetLatencyMs = ceil(audio.playback.jitter);
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int targetLatencyMs = 70;
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int targetLatencyFrames =
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targetLatencyMs * audio.playback.sampleRate / 1000;
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@ -585,16 +565,6 @@ void audio_playbackData(uint8_t * data, size_t size)
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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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// if the ringbuffer is full calculate the jitter
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if (ringbuffer_getCount(audio.playback.timings) ==
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ringbuffer_getLength(audio.playback.timings))
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{
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float minMax[2];
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ringbuffer_forEach(audio.playback.timings, getMinMax, minMax, false);
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audio.playback.jitter = minMax[1] - minMax[0];
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}
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app_invalidateGraph(audio.playback.graph);
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}
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