Remove unused members from WebRtcVoiceEngine and channels This cleanup removes member variables that were declared but never used in the implementation, likely as a result of previous refactorings or the send/receive channel split. - WebRtcVoiceEngine: removes is_dumping_aec_ - WebRtcVoiceSendChannel: removes unsignaled_frame_transformer_ - WebRtcVoiceReceiveChannel: removes send_codec_spec_ Bug: None Change-Id: I555ceca1b72f8d0f1d9e79b92945fda6850fc2bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47091}
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h index 0100fc4..97907fd 100644 --- a/media/engine/webrtc_voice_engine.h +++ b/media/engine/webrtc_voice_engine.h
@@ -162,7 +162,6 @@ scoped_refptr<AudioState> audio_state_ RTC_GUARDED_BY(worker_thread_checker_); const std::vector<Codec> legacy_send_codecs_; const std::vector<Codec> legacy_recv_codecs_; - bool is_dumping_aec_ RTC_GUARDED_BY(worker_thread_checker_) = false; bool initialized_ RTC_GUARDED_BY(worker_thread_checker_) = false; // Jitter buffer settings for new streams. @@ -304,8 +303,6 @@ // Per peer connection crypto options that last for the lifetime of the peer // connection. const CryptoOptions crypto_options_; - scoped_refptr<FrameTransformerInterface> unsignaled_frame_transformer_ - RTC_GUARDED_BY(worker_thread_); // Callback invoked whenever the list of SSRCs changes. absl::AnyInvocable<void(const std::set<uint32_t>&)> @@ -466,9 +463,6 @@ AudioReceiverParameters recv_params_ RTC_GUARDED_BY(worker_thread_); - std::optional<AudioSendStream::Config::SendCodecSpec> send_codec_spec_ - RTC_GUARDED_BY(worker_thread_); - // Per peer connection crypto options that last for the lifetime of the peer // connection. const CryptoOptions crypto_options_;