forked from mirrors/gecko-dev
3e423bd84b
Upstream commit: https://webrtc.googlesource.com/src/+/87e74f9fb77af8eba4f0b41e1441fd2b11261621 Remove unused combined_audio_video_bwe. Bug: None Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com> Cr-Commit-Position: refs/heads/main@{#40160}
75 lines
3 KiB
C++
75 lines
3 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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#ifndef API_AUDIO_OPTIONS_H_
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#define API_AUDIO_OPTIONS_H_
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#include <stdint.h>
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#include <string>
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#include "absl/types/optional.h"
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#include "rtc_base/system/rtc_export.h"
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namespace cricket {
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// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
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// Used to be flags, but that makes it hard to selectively apply options.
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// We are moving all of the setting of options to structs like this,
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// but some things currently still use flags.
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struct RTC_EXPORT AudioOptions {
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AudioOptions();
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~AudioOptions();
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void SetAll(const AudioOptions& change);
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bool operator==(const AudioOptions& o) const;
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bool operator!=(const AudioOptions& o) const { return !(*this == o); }
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std::string ToString() const;
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// Audio processing that attempts to filter away the output signal from
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// later inbound pickup.
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absl::optional<bool> echo_cancellation;
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#if defined(WEBRTC_IOS)
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// Forces software echo cancellation on iOS. This is a temporary workaround
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// (until Apple fixes the bug) for a device with non-functioning AEC. May
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// improve performance on that particular device, but will cause unpredictable
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// behavior in all other cases. See http://bugs.webrtc.org/8682.
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absl::optional<bool> ios_force_software_aec_HACK;
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#endif
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// Audio processing to adjust the sensitivity of the local mic dynamically.
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absl::optional<bool> auto_gain_control;
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// Audio processing to filter out background noise.
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absl::optional<bool> noise_suppression;
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// Audio processing to remove background noise of lower frequencies.
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absl::optional<bool> highpass_filter;
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// Audio processing to swap the left and right channels.
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absl::optional<bool> stereo_swapping;
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// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
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absl::optional<int> audio_jitter_buffer_max_packets;
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// Audio receiver jitter buffer (NetEq) fast accelerate mode.
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absl::optional<bool> audio_jitter_buffer_fast_accelerate;
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// Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
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absl::optional<int> audio_jitter_buffer_min_delay_ms;
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// Enable audio network adaptor.
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// TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
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// RtpEncodingParameters.
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absl::optional<bool> audio_network_adaptor;
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// Config string for audio network adaptor.
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absl::optional<std::string> audio_network_adaptor_config;
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// Pre-initialize the ADM for recording when starting to send. Default to
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// true.
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// TODO(webrtc:13566): Remove this option. See issue for details.
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absl::optional<bool> init_recording_on_send;
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};
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} // namespace cricket
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#endif // API_AUDIO_OPTIONS_H_
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