Index: webrtc/media/base/mediachannel.h |
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h |
index b71a309233546f941150431bf5df4aca8f46b095..a375d00a6215b745096589a4f8bf7d658a5a6f50 100644 |
--- a/webrtc/media/base/mediachannel.h |
+++ b/webrtc/media/base/mediachannel.h |
@@ -166,6 +166,7 @@ struct AudioOptions { |
SetFrom(&recording_sample_rate, change.recording_sample_rate); |
SetFrom(&playout_sample_rate, change.playout_sample_rate); |
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe); |
+ SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config); |
} |
bool operator==(const AudioOptions& o) const { |
@@ -193,7 +194,8 @@ struct AudioOptions { |
tx_agc_limiter == o.tx_agc_limiter && |
recording_sample_rate == o.recording_sample_rate && |
playout_sample_rate == o.playout_sample_rate && |
- combined_audio_video_bwe == o.combined_audio_video_bwe; |
+ combined_audio_video_bwe == o.combined_audio_video_bwe && |
+ audio_network_adaptor_config == o.audio_network_adaptor_config; |
} |
bool operator!=(const AudioOptions& o) const { return !(*this == o); } |
@@ -225,6 +227,8 @@ struct AudioOptions { |
ost << ToStringIfSet("recording_sample_rate", recording_sample_rate); |
ost << ToStringIfSet("playout_sample_rate", playout_sample_rate); |
ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe); |
+ ost << ToStringIfSet("audio_network_adaptor_config", |
+ audio_network_adaptor_config); |
ost << "}"; |
return ost.str(); |
} |
@@ -265,6 +269,8 @@ struct AudioOptions { |
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it, |
// and check if any other AudioOptions members are unused. |
rtc::Optional<bool> combined_audio_video_bwe; |
+ // Enable and config string for audio network adaptor. |
+ rtc::Optional<std::string> audio_network_adaptor_config; |
private: |
template <typename T> |