diff --git a/src/Limelight-internal.h b/src/Limelight-internal.h
index 10bcb8d..c8b77ab 100644
--- a/src/Limelight-internal.h
+++ b/src/Limelight-internal.h
@@ -37,6 +37,10 @@ extern int OriginalVideoBitrate;
 
 #define UDP_RECV_POLL_TIMEOUT_MS 100
 
+// At this value, we will request high quality audio unless CAPABILITY_SLOW_OPUS_DECODER
+// is set on the audio renderer.
+#define HIGH_AUDIO_BITRATE_THRESHOLD 15000
+
 int serviceEnetHost(ENetHost* client, ENetEvent* event, enet_uint32 timeoutMs);
 int extractVersionQuadFromString(const char* string, int* quad);
 int isReferenceFrameInvalidationEnabled(void);
diff --git a/src/Limelight.h b/src/Limelight.h
index 74bfad9..fdd065b 100644
--- a/src/Limelight.h
+++ b/src/Limelight.h
@@ -166,6 +166,11 @@ typedef struct _DECODE_UNIT {
 // supports reference frame invalidation for HEVC/H.265 streams. This flag is only valid on video renderers.
 #define CAPABILITY_REFERENCE_FRAME_INVALIDATION_HEVC 0x4
 
+// If set in the audio renderer capabilities field, this flag will cause the RTSP negotiation
+// to never request the "high quality" audio preset. If unset, high quality audio will be
+// used with video streams above 15 Mbps.
+#define CAPABILITY_SLOW_OPUS_DECODER 0x8
+
 // 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.
diff --git a/src/RtspConnection.c b/src/RtspConnection.c
index 8451d95..75d9139 100644
--- a/src/RtspConnection.c
+++ b/src/RtspConnection.c
@@ -430,9 +430,20 @@ static int sendVideoAnnounce(PRTSP_MESSAGE response, int* error) {
 int performRtspHandshake(void) {
     int ret;
 
+    // HACK: In order to get GFE to respect our request for a lower audio bitrate, we must
+    // fake our target address so it doesn't match any of the PC's local interfaces. It seems
+    // that the only way to get it to give you "low quality" stereo audio nowadays is if it
+    // thinks you are remote (target address != any local address).
+    if (OriginalVideoBitrate >= HIGH_AUDIO_BITRATE_THRESHOLD &&
+            (AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) == 0) {
+        addrToUrlSafeString(&RemoteAddr, urlAddr);
+    }
+    else {
+        strcpy(urlAddr, "0.0.0.0");
+    }
+
     // Initialize global state
     useEnet = (AppVersionQuad[0] >= 5) && (AppVersionQuad[0] <= 7) && (AppVersionQuad[2] < 404);
-    addrToUrlSafeString(&RemoteAddr, urlAddr);
     sprintf(rtspTargetUrl, "rtsp%s://%s:48010", useEnet ? "ru" : "", urlAddr);
     currentSeqNumber = 1;
     hasSessionId = 0;
diff --git a/src/SdpGenerator.c b/src/SdpGenerator.c
index 9bb085b..2019585 100644
--- a/src/SdpGenerator.c
+++ b/src/SdpGenerator.c
@@ -11,8 +11,6 @@
 #define CHANNEL_MASK_STEREO 0x3
 #define CHANNEL_MASK_51_SURROUND 0xFC
 
-#define HIGH_BITRATE_THRESHOLD 15000
-
 typedef struct _SDP_OPTION {
     char name[MAX_OPTION_NAME_LEN + 1];
     void* payload;
@@ -362,7 +360,8 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
 
         if (AppVersionQuad[0] >= 7) {
             // Decide to use HQ audio based on the original video bitrate, not the HEVC-adjusted value
-            if (OriginalVideoBitrate >= HIGH_BITRATE_THRESHOLD && audioChannelCount > 2) {
+            if (OriginalVideoBitrate >= HIGH_AUDIO_BITRATE_THRESHOLD && audioChannelCount > 2 &&
+                    (AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) == 0) {
                 // Enable high quality mode for surround sound
                 err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "1");