From 59481c085a7f774c5d30374636f8bc75da7c676c Mon Sep 17 00:00:00 2001
From: Cameron Gutman <aicommander@gmail.com>
Date: Sat, 4 May 2019 15:33:48 -0700
Subject: [PATCH] Add support for requesting 20 ms audio frames for slow Opus
 decoders

---
 src/AudioStream.c        | 14 ++++++++------
 src/Connection.c         |  1 +
 src/Limelight-internal.h |  1 +
 src/Limelight.h          |  1 +
 src/SdpGenerator.c       | 19 +++++++++++++++++++
 5 files changed, 30 insertions(+), 6 deletions(-)

diff --git a/src/AudioStream.c b/src/AudioStream.c
index d8160fa..20a0479 100644
--- a/src/AudioStream.c
+++ b/src/AudioStream.c
@@ -161,7 +161,7 @@ static void ReceiveThreadProc(void* context) {
     PQUEUED_AUDIO_PACKET packet;
     int queueStatus;
     int useSelect;
-    int packetsToDrop = 100;
+    int packetsToDrop = 500 / AudioPacketDuration;
 
     packet = NULL;
 
@@ -321,17 +321,17 @@ void stopAudioStream(void) {
 
 int startAudioStream(void* audioContext, int arFlags) {
     int err;
-    POPUS_MULTISTREAM_CONFIGURATION chosenConfig;
+    OPUS_MULTISTREAM_CONFIGURATION chosenConfig;
 
     if (StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_STEREO) {
-        chosenConfig = &opusStereoConfig;
+        chosenConfig = opusStereoConfig;
     }
     else if (StreamConfig.audioConfiguration == AUDIO_CONFIGURATION_51_SURROUND) {
         if (HighQualitySurroundEnabled) {
-            chosenConfig = &opus51HighSurroundConfig;
+            chosenConfig = opus51HighSurroundConfig;
         }
         else {
-            chosenConfig = &opus51SurroundConfig;
+            chosenConfig = opus51SurroundConfig;
         }
     }
     else {
@@ -339,7 +339,9 @@ int startAudioStream(void* audioContext, int arFlags) {
         return -1;
     }
 
-    err = AudioCallbacks.init(StreamConfig.audioConfiguration, chosenConfig, audioContext, arFlags);
+    chosenConfig.samplesPerFrame = 48 * AudioPacketDuration;
+
+    err = AudioCallbacks.init(StreamConfig.audioConfiguration, &chosenConfig, audioContext, arFlags);
     if (err != 0) {
         return err;
     }
diff --git a/src/Connection.c b/src/Connection.c
index 3361087..5bf2991 100644
--- a/src/Connection.c
+++ b/src/Connection.c
@@ -20,6 +20,7 @@ int NegotiatedVideoFormat;
 volatile int ConnectionInterrupted;
 int HighQualitySurroundEnabled;
 int OriginalVideoBitrate;
+int AudioPacketDuration;
 
 // Connection stages
 static const char* stageNames[STAGE_MAX] = {
diff --git a/src/Limelight-internal.h b/src/Limelight-internal.h
index 4c14ed6..f706684 100644
--- a/src/Limelight-internal.h
+++ b/src/Limelight-internal.h
@@ -22,6 +22,7 @@ extern int NegotiatedVideoFormat;
 extern volatile int ConnectionInterrupted;
 extern int HighQualitySurroundEnabled;
 extern int OriginalVideoBitrate;
+extern int AudioPacketDuration;
 
 #ifndef UINT24_MAX
 #define UINT24_MAX 0xFFFFFF
diff --git a/src/Limelight.h b/src/Limelight.h
index fdd065b..7e40eb2 100644
--- a/src/Limelight.h
+++ b/src/Limelight.h
@@ -228,6 +228,7 @@ typedef struct _OPUS_MULTISTREAM_CONFIGURATION {
     int channelCount;
     int streams;
     int coupledStreams;
+    int samplesPerFrame;
     unsigned char mapping[6];
 } OPUS_MULTISTREAM_CONFIGURATION, *POPUS_MULTISTREAM_CONFIGURATION;
 
diff --git a/src/SdpGenerator.c b/src/SdpGenerator.c
index 2019585..cc48b76 100644
--- a/src/SdpGenerator.c
+++ b/src/SdpGenerator.c
@@ -368,11 +368,30 @@ static PSDP_OPTION getAttributesList(char*urlSafeAddr) {
                 // Let the audio stream code know that it needs to disable coupled streams when
                 // decoding this audio stream.
                 HighQualitySurroundEnabled = 1;
+
+                // Use 5 ms frames since we don't have a slow decoder
+                AudioPacketDuration = 5;
             }
             else {
                 err |= addAttributeString(&optionHead, "x-nv-audio.surround.AudioQuality", "0");
                 HighQualitySurroundEnabled = 0;
+
+                if ((AudioCallbacks.capabilities & CAPABILITY_SLOW_OPUS_DECODER) == 0) {
+                    // Use 5 ms packets by default for lowest latency
+                    AudioPacketDuration = 5;
+                }
+                else {
+                    // Use 20 ms packets for slow decoders to save CPU and bandwidth
+                    AudioPacketDuration = 20;
+                }
             }
+
+            sprintf(payloadStr, "%d", AudioPacketDuration);
+            err |= addAttributeString(&optionHead, "x-nv-aqos.packetDuration", payloadStr);
+        }
+        else {
+            // 5 ms duration for legacy servers
+            AudioPacketDuration = 5;
         }
     }