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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_options.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio_options.h')
-rw-r--r-- | third_party/libwebrtc/api/audio_options.h | 80 |
1 files changed, 80 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_options.h b/third_party/libwebrtc/api/audio_options.h new file mode 100644 index 0000000000..39ba3886ea --- /dev/null +++ b/third_party/libwebrtc/api/audio_options.h @@ -0,0 +1,80 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_OPTIONS_H_ +#define API_AUDIO_OPTIONS_H_ + +#include <stdint.h> + +#include <string> + +#include "absl/types/optional.h" +#include "rtc_base/system/rtc_export.h" + +namespace cricket { + +// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine. +// Used to be flags, but that makes it hard to selectively apply options. +// We are moving all of the setting of options to structs like this, +// but some things currently still use flags. +struct RTC_EXPORT AudioOptions { + AudioOptions(); + ~AudioOptions(); + void SetAll(const AudioOptions& change); + + bool operator==(const AudioOptions& o) const; + bool operator!=(const AudioOptions& o) const { return !(*this == o); } + + std::string ToString() const; + + // Audio processing that attempts to filter away the output signal from + // later inbound pickup. + absl::optional<bool> echo_cancellation; +#if defined(WEBRTC_IOS) + // Forces software echo cancellation on iOS. This is a temporary workaround + // (until Apple fixes the bug) for a device with non-functioning AEC. May + // improve performance on that particular device, but will cause unpredictable + // behavior in all other cases. See http://bugs.webrtc.org/8682. + absl::optional<bool> ios_force_software_aec_HACK; +#endif + // Audio processing to adjust the sensitivity of the local mic dynamically. + absl::optional<bool> auto_gain_control; + // Audio processing to filter out background noise. + absl::optional<bool> noise_suppression; + // Audio processing to remove background noise of lower frequencies. + absl::optional<bool> highpass_filter; + // Audio processing to swap the left and right channels. + absl::optional<bool> stereo_swapping; + // Audio receiver jitter buffer (NetEq) max capacity in number of packets. + absl::optional<int> audio_jitter_buffer_max_packets; + // Audio receiver jitter buffer (NetEq) fast accelerate mode. + absl::optional<bool> audio_jitter_buffer_fast_accelerate; + // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds. + absl::optional<int> audio_jitter_buffer_min_delay_ms; + // Enable combined audio+bandwidth BWE. + // TODO(pthatcher): This flag is set from the + // "googCombinedAudioVideoBwe", but not used anywhere. So delete it, + // and check if any other AudioOptions members are unused. + absl::optional<bool> combined_audio_video_bwe; + // Enable audio network adaptor. + // TODO(webrtc:11717): Remove this API in favor of adaptivePtime in + // RtpEncodingParameters. + absl::optional<bool> audio_network_adaptor; + // Config string for audio network adaptor. + absl::optional<std::string> audio_network_adaptor_config; + // Pre-initialize the ADM for recording when starting to send. Default to + // true. + // TODO(webrtc:13566): Remove this option. See issue for details. + absl::optional<bool> init_recording_on_send; +}; + +} // namespace cricket + +#endif // API_AUDIO_OPTIONS_H_ |