Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 9f7ea58d90aa185edd14c4fdefd8708471bd4523..161cfe0643f6cef0add0dea08e820f872677476e 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1455,6 +1455,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
RTC_DCHECK(!stream_); |
if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") == |
"Enabled") { |
+ config_.min_bitrate_bps = kOpusMinBitrateBps; |
+ config_.max_bitrate_bps = kOpusBitrateFbBps; |
// TODO(mflodman): Keep testing this and set proper values. |
// Note: This is an early experiment currently only supported by Opus. |
if (webrtc::field_trial::FindFullName( |
@@ -1487,9 +1489,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream |
config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps; |
config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps; |
} |
- } else { |
- config_.min_bitrate_bps = kOpusMinBitrateBps; |
minyue-webrtc
2017/01/24 14:22:00
Then it should be outside if(webrtc::field_trial::
stefan-webrtc
2017/01/24 14:48:03
I don't follow, I moved it from the else branch to
|
- config_.max_bitrate_bps = kOpusBitrateFbBps; |
} |
} |
stream_ = call_->CreateAudioSendStream(config_); |