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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/api/audio_options.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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+/*
+ * 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_