Properly fit the rolling buffers to fit the set of sample frames

awslabs · Dec 4, 2024 · 2c3a466 · 2c3a466
1 parent 8c49c94
commit 2c3a466
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Showing 3 changed files with 24 additions and 0 deletions.
diff --git a/samples/Common.c b/samples/Common.c
@@ -532,6 +532,9 @@ STATUS createSampleStreamingSession(PSampleConfiguration pSampleConfiguration, P
     CHK_STATUS(addTransceiver(pSampleStreamingSession->pPeerConnection, &videoTrack, &videoRtpTransceiverInit,
                               &pSampleStreamingSession->pVideoRtcRtpTransceiver));
 
+    CHK_STATUS(configureTransceiverRollingBuffer(pSampleStreamingSession->pVideoRtcRtpTransceiver, &videoTrack,
+                                      pSampleConfiguration->videoRollingBufferDurationSec, pSampleConfiguration->videoRollingBufferBitratebps));
+
     CHK_STATUS(transceiverOnBandwidthEstimation(pSampleStreamingSession->pVideoRtcRtpTransceiver, (UINT64) pSampleStreamingSession,
                                                 sampleBandwidthEstimationHandler));
 
@@ -544,6 +547,9 @@ STATUS createSampleStreamingSession(PSampleConfiguration pSampleConfiguration, P
     CHK_STATUS(addTransceiver(pSampleStreamingSession->pPeerConnection, &audioTrack, &audioRtpTransceiverInit,
                               &pSampleStreamingSession->pAudioRtcRtpTransceiver));
 
+    CHK_STATUS(configureTransceiverRollingBuffer(pSampleStreamingSession->pAudioRtcRtpTransceiver, &audioTrack,
+                                      pSampleConfiguration->audioRollingBufferDurationSec, pSampleConfiguration->audioRollingBufferBitratebps));
+
     CHK_STATUS(transceiverOnBandwidthEstimation(pSampleStreamingSession->pAudioRtcRtpTransceiver, (UINT64) pSampleStreamingSession,
                                                 sampleBandwidthEstimationHandler));
     // twcc bandwidth estimation

diff --git a/samples/Samples.h b/samples/Samples.h
@@ -128,6 +128,10 @@ typedef struct {
     SIGNALING_CLIENT_HANDLE signalingClientHandle;
     RTC_CODEC audioCodec;
     RTC_CODEC videoCodec;
+    DOUBLE videoRollingBufferDurationSec;
+    DOUBLE videoRollingBufferBitratebps;
+    DOUBLE audioRollingBufferDurationSec;
+    DOUBLE audioRollingBufferBitratebps;
     PBYTE pAudioFrameBuffer;
     UINT32 audioBufferSize;
     PBYTE pVideoFrameBuffer;

diff --git a/samples/kvsWebRTCClientMaster.c b/samples/kvsWebRTCClientMaster.c
@@ -50,6 +50,20 @@ INT32 main(INT32 argc, CHAR* argv[])
     pSampleConfiguration->audioCodec = audioCodec;
     pSampleConfiguration->videoCodec = videoCodec;
 
+    // Configure the RTP rolling buffer sizes for the set of pre-canned sample frames (add a bit extra for padding)
+    if (pSampleConfiguration->videoCodec == RTC_CODEC_H264_PROFILE_42E01F_LEVEL_ASYMMETRY_ALLOWED_PACKETIZATION_MODE) {
+        pSampleConfiguration->videoRollingBufferDurationSec = 3;
+        pSampleConfiguration->videoRollingBufferBitratebps = 1.4 * 1024 * 1024;
+    } else if (pSampleConfiguration->videoCodec == RTC_CODEC_H265) {
+        pSampleConfiguration->videoRollingBufferDurationSec = 3;
+        pSampleConfiguration->videoRollingBufferBitratebps = 462 * 1024;
+    }
+
+    if (pSampleConfiguration->audioCodec == RTC_CODEC_OPUS) {
+        pSampleConfiguration->audioRollingBufferDurationSec = 3;
+        pSampleConfiguration->audioRollingBufferBitratebps = 512 * 1024;
+    }
+
     if (argc > 2 && STRNCMP(argv[2], "1", 2) == 0) {
         pSampleConfiguration->channelInfo.useMediaStorage = TRUE;
     }