Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(367)

Unified Diff: webrtc/video/video_quality_test.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index c5db0534c62f9ecb61bb261ce03547191f93e744..5fcfd193229c0a04369773105b38a63f703ab9ce 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1332,8 +1332,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
- audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
- audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
+ audio_send_config_.min_bitrate_bps = kOpusMinBitrate;
+ audio_send_config_.max_bitrate_bps = kOpusBitrateFb;
}
audio_send_config_.send_codec_spec.codec_inst =
CodecInst{120, "OPUS", 48000, 960, 2, 64000};
« webrtc/media/engine/webrtcvoiceengine.cc ('K') | « webrtc/media/engine/webrtcvoiceengine.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698