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author | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-11-18 12:18:43 +0000 |
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committer | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-11-18 12:18:43 +0000 |
commit | 4590177aa3346a742e8f71c8bcc5b81618d21919 (patch) | |
tree | e6de019bc9ecbe883aa6626b93d7107d8a12b5c9 /video_send_stream.h | |
parent | 09f84e54ee1e5ee952f573781e760f552aed2005 (diff) | |
download | webrtc-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.h | 10 |
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 |