| 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>
|
|
|