Rename AutoMute to SuspendBelowMinBitrate
Changes all instances throughout the WebRTC stack.
BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 2df282b..ee5826d 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -79,7 +79,7 @@
target_delay_ms(0),
pacing(false),
stats_callback(NULL),
- auto_mute(false) {}
+ suspend_below_min_bitrate(false) {}
VideoCodec codec;
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
@@ -142,10 +142,10 @@
// Callback for periodically receiving send stats.
StatsCallback* stats_callback;
- // True if video should be muted when video goes under the minimum video
- // bitrate. Unless muted, video will be sent at a bitrate higher than
- // estimated available.
- bool auto_mute;
+ // True if the stream should be suspended when the available bitrate fall
+ // below the minimum configured bitrate. If this variable is false, the
+ // stream may send at a rate higher than the estimated available bitrate.
+ bool suspend_below_min_bitrate;
};
// Gets interface used to insert captured frames. Valid as long as the
