Add support for 5.1 surround sound with GFE 2.7

limelight-common/AudioStream.c

@@ -17,13 +17,36 @@ static unsigned short lastSeq;
 
 #define RTP_PORT 48000
 
-#define MAX_PACKET_SIZE 100
+#define MAX_PACKET_SIZE 250
 
 // This is much larger than we should typically have buffered, but
 // it needs to be. We need a cushion in case our thread gets blocked
 // for longer than normal.
 #define RTP_RECV_BUFFER (64 * MAX_PACKET_SIZE)
 
+#define SAMPLE_RATE 48000
+
+static OPUS_MULTISTREAM_CONFIGURATION opusStereoConfig = {
+    .sampleRate = SAMPLE_RATE,
+    .channelCount = 2,
+    .streams = 1,
+    .coupledStreams = 1,
+    .mapping = {0, 1}
+};
+
+static OPUS_MULTISTREAM_CONFIGURATION opus51SurroundConfig = {
+    .sampleRate = SAMPLE_RATE,
+    .channelCount = 6,
+    .streams = 4,
+    .coupledStreams = 2,
+    .mapping = {0, 4, 1, 5, 2, 3}
+};
+
+static POPUS_MULTISTREAM_CONFIGURATION opusConfigArray[] = {
+    &opusStereoConfig,
+    &opus51SurroundConfig,
+};
+
 typedef struct _QUEUED_AUDIO_PACKET {
 	// data must remain at the front
 	char data[MAX_PACKET_SIZE];
@@ -242,7 +265,8 @@ void stopAudioStream(void) {
 int startAudioStream(void) {
 	int err;
 
-    AudioCallbacks.init();
+    AudioCallbacks.init(StreamConfig.audioConfiguration,
+        opusConfigArray[StreamConfig.audioConfiguration]);
 
 	rtpSocket = bindUdpSocket(RemoteAddr.ss_family, RTP_RECV_BUFFER);
 	if (rtpSocket == INVALID_SOCKET) {

limelight-common/FakeCallbacks.c

@@ -10,7 +10,7 @@ static DECODER_RENDERER_CALLBACKS fakeDrCallbacks = {
 	.submitDecodeUnit = fakeDrSubmitDecodeUnit,
 };
 
-static void fakeArInit(void) {}
+static void fakeArInit(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig) {}
 static void fakeArCleanup(void) {}
 static void fakeArDecodeAndPlaySample(char* sampleData, int sampleLength) {}
 

limelight-common/Limelight.h

@@ -29,6 +29,10 @@ typedef struct _STREAM_CONFIGURATION {
     // streaming optimizations. If unsure, set to 0.
     int streamingRemotely;
 
+    // Specifies the channel configuration of the audio stream.
+    // See AUDIO_CONFIGURATION_XXX constants below.
+    int audioConfiguration;
+
 	// AES encryption data for the remote input stream. This must be
 	// the same as what was passed as rikey and rikeyid 
 	// in /launch and /resume requests.
@@ -59,6 +63,12 @@ typedef struct _DECODE_UNIT {
 	PLENTRY bufferList;
 } DECODE_UNIT, *PDECODE_UNIT;
 
+// Specifies that the audio stream should be encoded in stereo (default)
+#define AUDIO_CONFIGURATION_STEREO 0
+
+// Specifies that the audio stream should be in 5.1 surround sound if the PC is able
+#define AUDIO_CONFIGURATION_51_SURROUND 1
+
 // If set in the renderer capabilities field, this flag will cause audio/video data to
 // be submitted directly from the receive thread. This should only be specified if the
 // renderer is non-blocking. This flag is valid on both audio and video renderers.
@@ -97,8 +107,21 @@ typedef struct _DECODER_RENDERER_CALLBACKS {
 // Use this function to zero the video callbacks when allocated on the stack or heap
 void LiInitializeVideoCallbacks(PDECODER_RENDERER_CALLBACKS drCallbacks);
 
-// This callback initializes the audio renderer
+// This structure provides the Opus multistream decoder parameters required to successfully
-typedef void(*AudioRendererInit)(void);
+// decode the audio stream being sent from the computer. See opus_multistream_decoder_init docs
+// for details about these fields.
+typedef struct _OPUS_MULTISTREAM_CONFIGURATION {
+    int sampleRate;
+    int channelCount;
+    int streams;
+    int coupledStreams;
+    const unsigned char mapping[6];
+} OPUS_MULTISTREAM_CONFIGURATION, *POPUS_MULTISTREAM_CONFIGURATION;
+
+// This callback initializes the audio renderer. The audio configuration parameter
+// provides the negotiated audio configuration. This may differ from the one
+// specified in the stream configuration.
+typedef void(*AudioRendererInit)(int audioConfiguration, POPUS_MULTISTREAM_CONFIGURATION opusConfig);
 
 // This callback performs the final teardown of the audio decoder
 typedef void(*AudioRendererCleanup)(void);

limelight-common/SdpGenerator.c

@@ -5,6 +5,12 @@
 #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;
@@ -134,6 +140,8 @@ static int addGen4Options(PSDP_OPTION *head, char* addrStr) {
     char payloadStr[92];
     int err = 0;
     unsigned char slicesPerFrame;
+    int audioChannelCount;
+    int audioChannelMask;
     
     sprintf(payloadStr, "rtsp://%s:48010", addrStr);
     err |= addAttributeString(head, "x-nv-general.serverAddress", payloadStr);
@@ -149,6 +157,26 @@ static int addGen4Options(PSDP_OPTION *head, char* addrStr) {
     sprintf(payloadStr, "%d", slicesPerFrame);
     err |= addAttributeString(head, "x-nv-video[0].videoEncoderSlicesPerFrame", payloadStr);
 
+    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;
+    }
+
+    sprintf(payloadStr, "%d", audioChannelCount);
+    err |= addAttributeString(head, "x-nv-audio.surround.numChannels", payloadStr);
+    sprintf(payloadStr, "%d", audioChannelMask);
+    err |= addAttributeString(head, "x-nv-audio.surround.channelMask", payloadStr);
+    if (audioChannelCount > 2) {
+        err |= addAttributeString(head, "x-nv-audio.surround.enable", "1");
+    }
+    else {
+        err |= addAttributeString(head, "x-nv-audio.surround.enable", "0");
+    }
+    
     return err;
 }