- ongoing work to fix issue with unstable audio
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@@ -60,16 +60,22 @@ namespace MT
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unsigned ssrc();
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void setSsrc(unsigned ssrc);
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void setHigh(int milliseconds);
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int high();
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void setLow(int milliseconds);
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int low();
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void setPrebuffer(int milliseconds);
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int prebuffer();
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int getNumberOfReturnedPackets() const;
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int getNumberOfAddPackets() const;
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int findTimelength();
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int getCount() const;
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// Returns false if packet was not add - maybe too old or too new or duplicate
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bool add(std::shared_ptr<RTPPacket> packet, int timelength, int rate);
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@@ -89,6 +95,9 @@ namespace MT
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bool mFirstPacketWillGo;
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jrtplib::RTPSourceStats mRtpStats;
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Packet mFetchedPacket;
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// To calculate average interval between packet add. It is close to jitter but more useful in debugging.
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float mLastAddTime = 0.0;
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};
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class Receiver
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@@ -167,6 +176,11 @@ namespace MT
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Audio::PWavFileWriter mDecodedDump;
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float mLastDecodeTime = 0.0; // Time last call happened to codec->decode()
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float mIntervalSum = 0.0;
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int mIntervalCount = 0;
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// Zero rate will make audio mono but resampling will be skipped
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void makeMonoAndResample(int rate, int channels);
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