| Index: webrtc/video/video_quality_test.cc
|
| diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
|
| index 565506c4b30130848e64243ccd4a36b85249c302..508ba13c836913c462ea89db2c3cb0fe2918550b 100644
|
| --- a/webrtc/video/video_quality_test.cc
|
| +++ b/webrtc/video/video_quality_test.cc
|
| @@ -1728,9 +1728,13 @@ void VideoQualityTest::SetupAudio(int send_channel_id,
|
| audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
|
| audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
|
| }
|
| - audio_send_config_.send_codec_spec.codec_inst =
|
| - CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
|
| - audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx;
|
| + audio_send_config_.send_codec_spec =
|
| + rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
|
| + {kAudioSendPayloadType,
|
| + {"OPUS", 48000, 2,
|
| + {{"usedtx", (params_.audio.dtx ? "1" : "0")},
|
| + {"stereo", "1"}}}});
|
| +
|
| audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
|
|
|
| AudioReceiveStream::Config audio_config;
|
|
|