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[client] audio/pw: limit how much data gets buffered to reduce latency
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@ -154,7 +154,7 @@ static void pipewire_start(int channels, int sampleRate)
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pw.channels = channels;
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pw.stride = sizeof(uint16_t) * channels;
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pw.buffer = ringbuffer_new(sampleRate, channels * sizeof(uint16_t));
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pw.buffer = ringbuffer_new(sampleRate / 10, channels * sizeof(uint16_t));
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pw_thread_loop_lock(pw.thread);
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pw.stream = pw_stream_new_simple(
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@ -203,6 +203,17 @@ static void pipewire_play(uint8_t * data, int size)
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if (!pw.stream)
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return;
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// if the buffer fill is higher then the average skip the update to reduce lag
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static unsigned int ttlSize = 0;
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static unsigned int count = 0;
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ttlSize += size;
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if (++count > 100 && ringbuffer_getCount(pw.buffer) > ttlSize / count)
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{
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count = 0;
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ttlSize = 0;
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return;
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}
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ringbuffer_append(pw.buffer, data, size / pw.stride);
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if (!pw.active)
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