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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_
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