Index: webrtc/media/base/mediachannel.h |
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h |
index 781a46d90c63122bc6504ca02f97cf3ecefa742a..e57cab1c7a0bba5a5bfa698b5a910ef114b34dcc 100644 |
--- a/webrtc/media/base/mediachannel.h |
+++ b/webrtc/media/base/mediachannel.h |
@@ -168,6 +168,7 @@ 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 { |
@@ -195,7 +196,8 @@ 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); } |
@@ -227,6 +229,8 @@ 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(); |
} |
@@ -267,6 +271,8 @@ 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 audio network adaptor and its config string. |
+ rtc::Optional<std::string> audio_network_adaptor_config; |
private: |
template <typename T> |