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authorhenrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-11-18 12:18:43 +0000
committerhenrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>2013-11-18 12:18:43 +0000
commit4590177aa3346a742e8f71c8bcc5b81618d21919 (patch)
treee6de019bc9ecbe883aa6626b93d7107d8a12b5c9 /video_send_stream.h
parent09f84e54ee1e5ee952f573781e760f552aed2005 (diff)
downloadwebrtc-4590177aa3346a742e8f71c8bcc5b81618d21919.tar.gz
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/webrtc@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'video_send_stream.h')
-rw-r--r--video_send_stream.h10
1 files changed, 5 insertions, 5 deletions
diff --git a/video_send_stream.h b/video_send_stream.h
index 2df282b4..ee5826dc 100644
--- a/video_send_stream.h
+++ b/video_send_stream.h
@@ -79,7 +79,7 @@ class VideoSendStream {
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 @@ class VideoSendStream {
// 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