diff --git a/src/Connection.c b/src/Connection.c
index 5dab9ed..740ef98 100644
--- a/src/Connection.c
+++ b/src/Connection.c
@@ -178,6 +178,14 @@ int LiStartConnection(PSERVER_INFORMATION serverInfo, PSTREAM_CONFIGURATION stre
     OriginalVideoBitrate = streamConfig->bitrate;
     RemoteAddrString = strdup(serverInfo->address);
     
+    // Validate the audio configuration
+    if (MAGIC_BYTE_FROM_AUDIO_CONFIG(StreamConfig.audioConfiguration) != 0xCA ||
+            CHANNEL_COUNT_FROM_AUDIO_CONFIGURATION(StreamConfig.audioConfiguration) > AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT) {
+        Limelog("Invalid audio configuration specified\n");
+        err = -1;
+        goto Cleanup;
+    }
+
     // FEC only works in 16 byte chunks, so we must round down
     // the given packet size to the nearest multiple of 16.
     StreamConfig.packetSize -= StreamConfig.packetSize % 16;
diff --git a/src/Limelight-internal.h b/src/Limelight-internal.h
index 8e32669..763aa7a 100644
--- a/src/Limelight-internal.h
+++ b/src/Limelight-internal.h
@@ -49,6 +49,9 @@ extern int AudioPacketDuration;
 // renderer sets CAPABILITY_SUPPORTS_ARBITRARY_AUDIO_DURATION.
 #define LOW_AUDIO_BITRATE_TRESHOLD 5000
 
+// Internal macro for checking the magic byte of the audio configuration value
+#define MAGIC_BYTE_FROM_AUDIO_CONFIG(x) ((x) & 0xFF)
+
 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 14bf980..dd54faa 100644
--- a/src/Limelight.h
+++ b/src/Limelight.h
@@ -52,7 +52,7 @@ typedef struct _STREAM_CONFIGURATION {
     int streamingRemotely;
 
     // Specifies the channel configuration of the audio stream.
-    // See AUDIO_CONFIGURATION_XXX constants below.
+    // See MAKE_AUDIO_CONFIGURATION() below.
     int audioConfiguration;
     
     // Specifies that the client can accept an H.265 video stream
@@ -153,10 +153,23 @@ typedef struct _DECODE_UNIT {
 } DECODE_UNIT, *PDECODE_UNIT;
 
 // Specifies that the audio stream should be encoded in stereo (default)
-#define AUDIO_CONFIGURATION_STEREO 0
+// Deprecated: use MAKE_AUDIO_CONFIGURATION() instead!
+#define AUDIO_CONFIGURATION_STEREO MAKE_AUDIO_CONFIGURATION(2, 0x3)
 
 // Specifies that the audio stream should be in 5.1 surround sound if the PC is able
-#define AUDIO_CONFIGURATION_51_SURROUND 1
+// Deprecated: use MAKE_AUDIO_CONFIGURATION() instead!
+#define AUDIO_CONFIGURATION_51_SURROUND MAKE_AUDIO_CONFIGURATION(6, 0xFC)
+
+// Specifies an audio configuration by channel count and channel mask
+#define MAKE_AUDIO_CONFIGURATION(channelCount, channelMask) \
+    (((channelMask) << 16) | (channelCount << 8) | 0xCA)
+
+// Helper macros for retreiving channel count and channel mask from the audio configuration
+#define CHANNEL_COUNT_FROM_AUDIO_CONFIGURATION(x) (((x) >> 8) & 0xFF)
+#define CHANNEL_MASK_FROM_AUDIO_CONFIGURATION(x) (((x) >> 16) & 0xFF)
+
+// The maximum number of channels supported
+#define AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT 6
 
 // Passed to DecoderRendererSetup to indicate that the following video stream will be
 // in H.264 High Profile.
@@ -258,7 +271,7 @@ typedef struct _OPUS_MULTISTREAM_CONFIGURATION {
     int streams;
     int coupledStreams;
     int samplesPerFrame;
-    unsigned char mapping[6];
+    unsigned char mapping[AUDIO_CONFIGURATION_MAX_CHANNEL_COUNT];
 } OPUS_MULTISTREAM_CONFIGURATION, *POPUS_MULTISTREAM_CONFIGURATION;
 
 // This callback initializes the audio renderer. The audio configuration parameter
diff --git a/src/RtspConnection.c b/src/RtspConnection.c
index 4943f90..3cbc11a 100644
--- a/src/RtspConnection.c
+++ b/src/RtspConnection.c
@@ -488,7 +488,7 @@ static int parseOpusConfigurations(PRTSP_MESSAGE response) {
     HighQualityOpusConfig.sampleRate = NormalQualityOpusConfig.sampleRate = 48000;
 
     // Stereo doesn't have any surround-params elements in the RTSP data
-    if (StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_STEREO) {
+    if (CHANNEL_COUNT_FROM_AUDIO_CONFIGURATION(StreamConfig.audioConfiguration) == 2) {
         NormalQualityOpusConfig.channelCount = 2;
         NormalQualityOpusConfig.streams = 1;
         NormalQualityOpusConfig.coupledStreams = 1;
@@ -501,8 +501,7 @@ static int parseOpusConfigurations(PRTSP_MESSAGE response) {
         int err;
         int channelCount;
 
-        LC_ASSERT(StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_51_SURROUND);
-        channelCount = 6;
+        channelCount = CHANNEL_COUNT_FROM_AUDIO_CONFIGURATION(StreamConfig.audioConfiguration);
 
         // Find the correct audio parameter value
         sprintf(paramsPrefix, "a=fmtp:97 surround-params=%d", channelCount);
@@ -548,7 +547,7 @@ static int parseOpusConfigurations(PRTSP_MESSAGE response) {
             // It's unknown whether all GFE versions that supported surround sound included these
             // surround sound parameters. In case they didn't, we'll specifically handle 5.1 surround
             // sound using a hardcoded configuration like we used to before this parsing code existed.
-            if (StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_51_SURROUND) {
+            if (channelCount == 6) {
                 NormalQualityOpusConfig.channelCount = 6;
                 NormalQualityOpusConfig.streams = 4;
                 NormalQualityOpusConfig.coupledStreams = 2;
diff --git a/src/SdpGenerator.c b/src/SdpGenerator.c
index ad958b9..699d246 100644
--- a/src/SdpGenerator.c
+++ b/src/SdpGenerator.c
@@ -5,12 +5,6 @@
 #define MAX_SDP_HEADER_LEN 128
 #define MAX_SDP_TAIL_LEN 128
 
-#define CHANNEL_COUNT_STEREO 2
-#define CHANNEL_COUNT_51_SURROUND 6
-
-#define CHANNEL_MASK_STEREO 0x3
-#define CHANNEL_MASK_51_SURROUND 0xFC
-
 typedef struct _SDP_OPTION {
     char name[MAX_OPTION_NAME_LEN + 1];
     void* payload;
@@ -266,14 +260,8 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
         err |= addGen5Options(&optionHead);
     }
 
-    if (StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_51_SURROUND) {
-        audioChannelCount = CHANNEL_COUNT_51_SURROUND;
-        audioChannelMask = CHANNEL_MASK_51_SURROUND;
-    }
-    else {
-        audioChannelCount = CHANNEL_COUNT_STEREO;
-        audioChannelMask = CHANNEL_MASK_STEREO;
-    }
+    audioChannelCount = CHANNEL_COUNT_FROM_AUDIO_CONFIGURATION(StreamConfig.audioConfiguration);
+    audioChannelMask = CHANNEL_MASK_FROM_AUDIO_CONFIGURATION(StreamConfig.audioConfiguration);
 
     if (AppVersionQuad[0] >= 4) {
         unsigned char slicesPerFrame;