| Index: webrtc/audio_send_stream.h
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| diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
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| index 806b3d4a3cc3258644119f693f144db021a4836d..ffb5f9c4d56595905c2c1c9e45f671694633dfb4 100644
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| --- a/webrtc/audio_send_stream.h
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| +++ b/webrtc/audio_send_stream.h
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| @@ -56,7 +56,7 @@ class AudioSendStream {
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|  
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|      std::string ToString() const;
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|  
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| -    // Receive-stream specific RTP settings.
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| +    // Send-stream specific RTP settings.
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|      struct Rtp {
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|        std::string ToString() const;
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|  
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| @@ -66,6 +66,9 @@ class AudioSendStream {
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|        // RTP header extensions used for the sent stream.
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|        std::vector<RtpExtension> extensions;
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|  
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| +      // See NackConfig for description.
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| +      NackConfig nack;
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| +
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|        // RTCP CNAME, see RFC 3550.
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|        std::string c_name;
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|      } rtp;
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| 
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