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a751f167c6
This reverts commit b4e06d032e.
Reason for revert: breaking downstream projects
Original change's description:
> Remove unused APM voice activity detection sub-module
>
> API changes:
> - webrtc::AudioProcessing::Config::VoiceDetection removed
> - webrtc::AudioProcessingStats::voice_detected deprecated
> - cricket::AudioOptions::typing_detection deprecated
> - webrtc::StatsReport::StatsValueName::
> kStatsValueNameTypingNoiseState deprecated
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35975}
TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
94 lines
3.9 KiB
C++
94 lines
3.9 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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// Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
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absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
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// Audio processing to detect typing.
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absl::optional<bool> typing_detection;
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absl::optional<bool> experimental_agc;
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absl::optional<bool> experimental_ns;
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// TODO(bugs.webrtc.org/11539): Deprecated, replaced by
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// webrtc::CreateEchoDetector() and injection when creating the audio
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// processing module.
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absl::optional<bool> residual_echo_detector;
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// Note that tx_agc_* only applies to non-experimental AGC.
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absl::optional<uint16_t> tx_agc_target_dbov;
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absl::optional<uint16_t> tx_agc_digital_compression_gain;
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absl::optional<bool> tx_agc_limiter;
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// Enable combined audio+bandwidth BWE.
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// TODO(pthatcher): This flag is set from the
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// "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
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// and check if any other AudioOptions members are unused.
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absl::optional<bool> combined_audio_video_bwe;
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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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