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Samples: Fit the RTP transceiver rolling buffers to fit the set of sample frames #2089

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Dec 5, 2024
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Samples: Fit the RTP transceiver rolling buffers to fit the set of sample frames #2089

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a355a19
Properly fit the rolling buffers to fit the set of sample frames
sirknightj Dec 4, 2024
b576787
Clang-format
sirknightj Dec 4, 2024
9fb148e
Adjust the comment for the rolling buffer configuration for the sampl…
sirknightj Dec 5, 2024
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8 changes: 8 additions & 0 deletions samples/Common.c
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Original file line number Diff line number Diff line change
Expand Up @@ -532,6 +532,10 @@ 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));

Expand All @@ -544,6 +548,10 @@ 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
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4 changes: 4 additions & 0 deletions samples/Samples.h
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Expand Up @@ -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;
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14 changes: 14 additions & 0 deletions samples/kvsWebRTCClientMaster.c
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Expand Up @@ -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)
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if (pSampleConfiguration->videoCodec == RTC_CODEC_H264_PROFILE_42E01F_LEVEL_ASYMMETRY_ALLOWED_PACKETIZATION_MODE) {
pSampleConfiguration->videoRollingBufferDurationSec = 3;
pSampleConfiguration->videoRollingBufferBitratebps = 1.4 * 1024 * 1024;
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Note: In the other samples this is not set, meaning default values are used. Since CALLOC is used for the sample configuration, these values are 0 everywhere else. Calling configureTransceiverRollingBuffer with 0's (or also not calling it at all) will use the default values.

} 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;
}
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