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[client] audio/pw: limit how much data gets buffered to reduce latency

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This commit is contained in:
Geoffrey McRae 2021-12-25 13:36:08 +11:00
parent dd04a46403
commit 2ea24516d2
1 changed files with 12 additions and 1 deletions
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13
client/audiodevs/PipeWire/pipewire.c
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@ -154,7 +154,7 @@ static void pipewire_start(int channels, int sampleRate)
pw.channels = channels;
pw.stride = sizeof(uint16_t) * channels;
pw.buffer = ringbuffer_new(sampleRate, channels * sizeof(uint16_t));
pw.buffer = ringbuffer_new(sampleRate / 10, channels * sizeof(uint16_t));
pw_thread_loop_lock(pw.thread);
pw.stream = pw_stream_new_simple(
@ -203,6 +203,17 @@ static void pipewire_play(uint8_t * data, int size)
if (!pw.stream)
return;
// if the buffer fill is higher then the average skip the update to reduce lag
static unsigned int ttlSize = 0;
static unsigned int count = 0;
ttlSize += size;
if (++count > 100 && ringbuffer_getCount(pw.buffer) > ttlSize / count)
{
count = 0;
ttlSize = 0;
return;
}
ringbuffer_append(pw.buffer, data, size / pw.stride);
if (!pw.active)