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Remove deprecated audio option residual_echo_detector

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The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ

Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36742}
This commit is contained in:
Sam Zackrisson
2022-04-26 11:36:12 +02:00
committed by WebRTC LUCI CQ
parent 3a9e6877e5
commit eeffb6aaf7
2 changed files with 0 additions and 7 deletions
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3
api/audio_options.cc
View File

@ -54,7 +54,6 @@ void AudioOptions::SetAll(const AudioOptions& change) {
change.audio_jitter_buffer_min_delay_ms);
SetFrom(&audio_jitter_buffer_enable_rtx_handling,
change.audio_jitter_buffer_enable_rtx_handling);
SetFrom(&residual_echo_detector, change.residual_echo_detector);
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
@ -77,7 +76,6 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
o.audio_jitter_buffer_min_delay_ms &&
audio_jitter_buffer_enable_rtx_handling ==
o.audio_jitter_buffer_enable_rtx_handling &&
residual_echo_detector == o.residual_echo_detector &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config &&
@ -105,7 +103,6 @@ std::string AudioOptions::ToString() const {
audio_jitter_buffer_min_delay_ms);
ToStringIfSet(&result, "audio_jitter_buffer_enable_rtx_handling",
audio_jitter_buffer_enable_rtx_handling);
ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);

4
api/audio_options.h
View File

@ -60,10 +60,6 @@ struct RTC_EXPORT AudioOptions {
absl::optional<int> audio_jitter_buffer_min_delay_ms;
// Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
// TODO(bugs.webrtc.org/11539): Deprecated, replaced by
// webrtc::CreateEchoDetector() and injection when creating the audio
// processing module.
absl::optional<bool> residual_echo_detector;
// Enable combined audio+bandwidth BWE.
// TODO(pthatcher): This flag is set from the
// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,