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552d3e3d5e
This is a reland of 8be2f201ba
Original change's description:
> Add ability to state whether the APM output will be used
>
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
>
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}
Bug: b/154437967
Bug: b/163802450
Change-Id: Ia77a9e43f913929d1afa72212f1ea6c192d0e519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181887
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31957}
50 lines
2 KiB
C++
50 lines
2 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/test/runtime_setting_util.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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void ReplayRuntimeSetting(AudioProcessing* apm,
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const webrtc::audioproc::RuntimeSetting& setting) {
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RTC_CHECK(apm);
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// TODO(bugs.webrtc.org/9138): Add ability to handle different types
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// of settings. Currently CapturePreGain, CaptureFixedPostGain and
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// PlayoutVolumeChange are supported.
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RTC_CHECK(setting.has_capture_pre_gain() ||
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setting.has_capture_fixed_post_gain() ||
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setting.has_playout_volume_change());
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if (setting.has_capture_pre_gain()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(
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setting.capture_pre_gain()));
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} else if (setting.has_capture_fixed_post_gain()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureFixedPostGain(
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setting.capture_fixed_post_gain()));
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} else if (setting.has_playout_volume_change()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(
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setting.playout_volume_change()));
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} else if (setting.has_playout_audio_device_change()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreatePlayoutAudioDeviceChange(
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{setting.playout_audio_device_change().id(),
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setting.playout_audio_device_change().max_volume()}));
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} else if (setting.has_capture_output_used()) {
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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setting.capture_output_used()));
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}
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}
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} // namespace webrtc
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