(Auto)update libjingle 75924589-> 75925673

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 8d8b36c..1ee3975 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -2884,7 +2884,7 @@
   }
 
   // On success, SetSendCodec() will reset |send_start_bitrate_| to |bps/1000|,
-  // by calling MaybeChangeBitrates.  That method will also clamp the
+  // by calling SanitizeBitrates.  That method will also clamp the
   // start bitrate between min and max, consistent with the override behavior
   // in SetMaxSendBandwidth.
   webrtc::VideoCodec new_codec = *send_codec_;
@@ -3659,7 +3659,7 @@
                  << "for ssrc: " << ssrc << ".";
   } else {
     StreamParams* send_params = send_channel->stream_params();
-    MaybeChangeBitrates(channel_id, &target_codec);
+    SanitizeBitrates(channel_id, &target_codec);
     webrtc::VideoCodec current_codec;
     if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) {
       // Compare against existing configured send codec.
@@ -3946,7 +3946,7 @@
       vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
       vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
     }
-    MaybeChangeBitrates(channel_id, &vie_codec);
+    SanitizeBitrates(channel_id, &vie_codec);
 
     if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) {
       LOG_RTCERR1(SetSendCodec, channel_id);
@@ -3984,7 +3984,7 @@
   return true;
 }
 
-void WebRtcVideoMediaChannel::MaybeChangeBitrates(
+void WebRtcVideoMediaChannel::SanitizeBitrates(
   int channel_id, webrtc::VideoCodec* codec) {
   codec->minBitrate = GetBitrate(codec->minBitrate, kMinVideoBitrate);
   codec->startBitrate = GetBitrate(codec->startBitrate, kStartVideoBitrate);
diff --git a/talk/media/webrtc/webrtcvideoengine.h b/talk/media/webrtc/webrtcvideoengine.h
index fb0c4e1..275f039 100644
--- a/talk/media/webrtc/webrtcvideoengine.h
+++ b/talk/media/webrtc/webrtcvideoengine.h
@@ -310,6 +310,10 @@
   virtual int SendPacket(int channel, const void* data, int len);
   virtual int SendRTCPPacket(int channel, const void* data, int len);
 
+  // Checks the current bitrate estimate and modifies the bitrates
+  // accordingly, including converting kAutoBandwidth to the correct defaults.
+  virtual void SanitizeBitrates(
+      int channel_id, webrtc::VideoCodec* video_codec);
   virtual void LogSendCodecChange(const std::string& reason);
   bool SetPrimaryAndRtxSsrcs(
       int channel_id, int idx, uint32 primary_ssrc,
@@ -361,9 +365,6 @@
   bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel,
                               int new_width, int new_height, bool is_screencast,
                               bool* reset);
-  // Checks the current bitrate estimate and modifies the bitrates
-  // accordingly, including converting kAutoBandwidth to the correct defaults.
-  void MaybeChangeBitrates(int channel_id, webrtc::VideoCodec* video_codec);
   // Helper function for starting the sending of media on all channels or
   // |channel_id|. Note that these two function do not change |sending_|.
   bool StartSend();