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authordec05eba <dec05eba@protonmail.com>2024-04-09 23:34:35 +0200
committerdec05eba <dec05eba@protonmail.com>2024-04-09 23:34:35 +0200
commite7aa4a549951a4989f2a1506f891540dc1bc72c1 (patch)
treedfc60ad96bed8f929a667a3bb9a6ae95e974af30 /src
parentd5bf41fed6628083200370cd5acf3ff43b592891 (diff)
Better audio timing test: compensate for audio server latency
Diffstat (limited to 'src')
-rw-r--r--src/main.cpp4
1 files changed, 4 insertions, 0 deletions
diff --git a/src/main.cpp b/src/main.cpp
index 167f8a5..bb54ca2 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -2388,6 +2388,10 @@ int main(int argc, char **argv) {
double received_audio_time = clock_get_monotonic_seconds();
const int64_t timeout_ms = std::round((1000.0 / (double)audio_track.codec_context->sample_rate) * 1000.0);
+ // Move audio back by around 252 ms. This is just a shitty way to handle audio latency but pulseaudio latency calculation
+ // returns much lower value which isn't helpful.
+ audio_device.frame->pts = audio_track.codec_context->frame_size * 12;
+
while(running) {
void *sound_buffer;
int sound_buffer_size = -1;