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Use 10 ms audio packets on low bitrate streams to balance latency and bandwidth

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This commit is contained in:
Cameron Gutman 2020-03-18 20:03:56 -07:00
commit c1b8aa266f
1 changed files with 7 additions and 5 deletions
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12
src/SdpGenerator.c
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@ -372,14 +372,16 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "0"); err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "0");
HighQualitySurroundEnabled = 0; HighQualitySurroundEnabled = 0;
if ((AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) != 0 || if ((AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) != 0) {
((AudioCallbacks.capabilities & CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION) != 0 && // Use 20 ms packets for slow decoders to save CPU time
OriginalVideoBitrate < LOW_AUDIO_BITRATE_TRESHOLD)) {
// Use 20 ms packets for slow decoders and networks to save CPU and bandwidth
AudioPacketDuration = 20; AudioPacketDuration = 20;
} }
else if ((AudioCallbacks.capabilities & CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION) != 0 &&
OriginalVideoBitrate < LOW_AUDIO_BITRATE_TRESHOLD) {
// Use 10 ms packets for slow networks to balance latency and bandwidth usage
AudioPacketDuration = 10;
}
else { else {
// Use 5 ms packets by default for lowest latency // Use 5 ms packets by default for lowest latency
AudioPacketDuration = 5; AudioPacketDuration = 5;
} }