obs-webrtc: Increase RtcpNackResponder size

Before we used the default value set by libdatachannel. At
higher bitrates the cache would be undersized. Because of the undersized
cache we wouldn't have enough NACK history to fix the stream.

plugins/obs-webrtc/whip-output.cpp

@@ -21,6 +21,9 @@ const uint8_t audio_payload_type = 111;
 const char *video_mid = "1";
 const uint8_t video_payload_type = 96;
 
+// ~3 seconds of 8.5 Megabit video
+const int video_nack_buffer_size = 4000;
+
 WHIPOutput::WHIPOutput(obs_data_t *, obs_output_t *output)
 	: output(output),
 	  endpoint_url(),
@@ -121,7 +124,8 @@ void WHIPOutput::ConfigureAudioTrack(std::string media_stream_id,
 		rtc::OpusRtpPacketizer::DefaultClockRate);
 	auto packetizer = std::make_shared<rtc::OpusRtpPacketizer>(rtp_config);
 	audio_sr_reporter = std::make_shared<rtc::RtcpSrReporter>(rtp_config);
-	auto nack_responder = std::make_shared<rtc::RtcpNackResponder>();
+	auto nack_responder = std::make_shared<rtc::RtcpNackResponder>(
+		video_nack_buffer_size);
 
 	packetizer->addToChain(audio_sr_reporter);
 	packetizer->addToChain(nack_responder);