webrtc/api/audio_options.h
Alessio Bazzica 54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c6.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> 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}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
2022-02-13 14:02:08 +00:00

96 lines
4 KiB
C++

/*
* 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;
// Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
// Deprecated.
// TODO(bugs.webrtc.org/11226): Remove.
// Audio processing to detect typing.
absl::optional<bool> typing_detection;
absl::optional<bool> experimental_agc;
absl::optional<bool> experimental_ns;
// TODO(bugs.webrtc.org/11539): Deprecated, replaced by
// webrtc::CreateEchoDetector() and injection when creating the audio
// processing module.
absl::optional<bool> residual_echo_detector;
// Note that tx_agc_* only applies to non-experimental AGC.
absl::optional<uint16_t> tx_agc_target_dbov;
absl::optional<uint16_t> tx_agc_digital_compression_gain;
absl::optional<bool> tx_agc_limiter;
// 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_