blob: df66d360c562359994adc03ef73d8ab8236aff88 [file] [log] [blame]
/*
* 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 <string>
#include "absl/types/optional.h"
#include "rtc_base/stringencode.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 AudioOptions {
AudioOptions();
~AudioOptions();
void SetAll(const AudioOptions& change) {
SetFrom(&echo_cancellation, change.echo_cancellation);
#if defined(WEBRTC_IOS)
SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
#endif
SetFrom(&auto_gain_control, change.auto_gain_control);
SetFrom(&noise_suppression, change.noise_suppression);
SetFrom(&highpass_filter, change.highpass_filter);
SetFrom(&stereo_swapping, change.stereo_swapping);
SetFrom(&audio_jitter_buffer_max_packets,
change.audio_jitter_buffer_max_packets);
SetFrom(&audio_jitter_buffer_fast_accelerate,
change.audio_jitter_buffer_fast_accelerate);
SetFrom(&typing_detection, change.typing_detection);
SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
SetFrom(&experimental_agc, change.experimental_agc);
SetFrom(&extended_filter_aec, change.extended_filter_aec);
SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
SetFrom(&experimental_ns, change.experimental_ns);
SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
SetFrom(&residual_echo_detector, change.residual_echo_detector);
SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
SetFrom(&tx_agc_digital_compression_gain,
change.tx_agc_digital_compression_gain);
SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
}
bool operator==(const AudioOptions& o) const {
return echo_cancellation == o.echo_cancellation &&
#if defined(WEBRTC_IOS)
ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
#endif
auto_gain_control == o.auto_gain_control &&
noise_suppression == o.noise_suppression &&
highpass_filter == o.highpass_filter &&
stereo_swapping == o.stereo_swapping &&
audio_jitter_buffer_max_packets ==
o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
typing_detection == o.typing_detection &&
aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
experimental_agc == o.experimental_agc &&
extended_filter_aec == o.extended_filter_aec &&
delay_agnostic_aec == o.delay_agnostic_aec &&
experimental_ns == o.experimental_ns &&
intelligibility_enhancer == o.intelligibility_enhancer &&
residual_echo_detector == o.residual_echo_detector &&
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain ==
o.tx_agc_digital_compression_gain &&
tx_agc_limiter == o.tx_agc_limiter &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
audio_network_adaptor_config == o.audio_network_adaptor_config;
}
bool operator!=(const AudioOptions& o) const { return !(*this == o); }
std::string ToString() const {
std::ostringstream ost;
ost << "AudioOptions {";
ost << ToStringIfSet("aec", echo_cancellation);
#if defined(WEBRTC_IOS)
ost << ToStringIfSet("ios_force_software_aec_HACK",
ios_force_software_aec_HACK);
#endif
ost << ToStringIfSet("agc", auto_gain_control);
ost << ToStringIfSet("ns", noise_suppression);
ost << ToStringIfSet("hf", highpass_filter);
ost << ToStringIfSet("swap", stereo_swapping);
ost << ToStringIfSet("audio_jitter_buffer_max_packets",
audio_jitter_buffer_max_packets);
ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
audio_jitter_buffer_fast_accelerate);
ost << ToStringIfSet("typing", typing_detection);
ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
ost << ToStringIfSet("experimental_agc", experimental_agc);
ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
ost << ToStringIfSet("experimental_ns", experimental_ns);
ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
ost << ToStringIfSet("tx_agc_digital_compression_gain",
tx_agc_digital_compression_gain);
ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
// The adaptor config is a serialized proto buffer and therefore not human
// readable. So we comment out the following line.
// ost << ToStringIfSet("audio_network_adaptor_config",
// audio_network_adaptor_config);
ost << "}";
return ost.str();
}
// 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 processing to detect typing.
absl::optional<bool> typing_detection;
absl::optional<bool> aecm_generate_comfort_noise;
absl::optional<bool> experimental_agc;
absl::optional<bool> extended_filter_aec;
absl::optional<bool> delay_agnostic_aec;
absl::optional<bool> experimental_ns;
absl::optional<bool> intelligibility_enhancer;
// Note that tx_agc_* only applies to non-experimental AGC.
absl::optional<bool> residual_echo_detector;
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.
absl::optional<bool> audio_network_adaptor;
// Config string for audio network adaptor.
absl::optional<std::string> audio_network_adaptor_config;
private:
template <class T>
static std::string ToStringIfSet(const char* key,
const absl::optional<T>& val) {
std::string str;
if (val) {
str = key;
str += ": ";
str += val ? rtc::ToString(*val) : "";
str += ", ";
}
return str;
}
template <typename T>
static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
if (o) {
*s = o;
}
}
};
} // namespace cricket
#endif // API_AUDIO_OPTIONS_H_