webrtc/media/engine/webrtcvoiceengine.cc - Issue 2247213005: Fixing config for Audio BWE.

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Unified Diff: webrtc/media/engine/webrtcvoiceengine.cc

Issue 2247213005: Fixing config for Audio BWE. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: rebasing Created 4 years, 2 months ago

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Index: webrtc/media/engine/webrtcvoiceengine.cc

diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc

index ebcd1613946cbf82b07fdbe1f1b36bffa885bcb4..2493d66ac8d0e4de41524e3f55d1b844a0baa258 100644

--- a/webrtc/media/engine/webrtcvoiceengine.cc

+++ b/webrtc/media/engine/webrtcvoiceengine.cc

@@ -1358,8 +1358,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream

"Enabled") {

// TODO(mflodman): Keep testing this and set proper values.

// Note: This is an early experiment currently only supported by Opus.

- config_.min_bitrate_kbps = kOpusMinBitrate;

- config_.max_bitrate_kbps = kOpusBitrateFb;

+ config_.min_bitrate_bps = kOpusMinBitrate;

+ config_.max_bitrate_bps = kOpusBitrateFb;

stefan-webrtc 2016/11/07 12:33:34 I would suggest adding units to the constants too.

}

stream_ = call_->CreateAudioSendStream(config_);

RTC_CHECK(stream_);

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