Increase audio packet duration for low bandwidth connections

src/Limelight-internal.h

@@ -39,10 +39,14 @@ extern int AudioPacketDuration;
 
 #define UDP_RECV_POLL_TIMEOUT_MS 100
 
-// At this value, we will request high quality audio unless CAPABILITY_SLOW_OPUS_DECODER
+// At this value or above, we will request high quality audio unless CAPABILITY_SLOW_OPUS_DECODER
 // is set on the audio renderer.
 #define HIGH_AUDIO_BITRATE_THRESHOLD 15000
 
+// Below this value, we will request 20 ms audio frames to reduce bandwidth if the audio
+// renderer sets CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION.
+#define LOW_AUDIO_BITRATE_TRESHOLD 5000
+
 int serviceEnetHost(ENetHost* client, ENetEvent* event, enet_uint32 timeoutMs);
 int extractVersionQuadFromString(const char* string, int* quad);
 int isReferenceFrameInvalidationEnabled(void);

src/Limelight.h

@@ -176,6 +176,12 @@ typedef struct _DECODE_UNIT {
 // used with video streams above 15 Mbps.
 #define CAPABILITY_SLOW_OPUS_DECODER 0x8
 
+// If set in the audio renderer capabilities field, this indicates that audio packets
+// may contain more or less than 5 ms of audio. This requires that audio renderers read the
+// samplesPerFrame field in OPUS_MULTISTREAM_CONFIGURATION to calculate the correct decoded
+// buffer size rather than just assuming it will always be 240.
+#define CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION 0x10
+
 // If set in the video renderer capabilities field, this macro specifies that the renderer
 // supports slicing to increase decoding performance. The parameter specifies the desired
 // number of slices per frame. This capability is only valid on video renderers.

src/SdpGenerator.c

@@ -377,7 +377,9 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
             err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "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 &&
+                    OriginalVideoBitrate >= LOW_AUDIO_BITRATE_TRESHOLD)) {
                 // Use 5 ms packets by default for lowest latency
                 AudioPacketDuration = 5;
             }