直接贴代码，说明都写在注释中了。
从下面代码中可以看出iOS平台的VPIO自己本身已经支持AEC、AGC和NS所以不使用WebRTC的软件算法。
在iOS平台可以通过ios_force_software_aec_HACK 强制开启软件回声消除:echo_cancellation/extended_filter_aec，
但是AGC和NS目前没有选项可以设置。
如果安卓平台内置了AEC、AGC和NS也是不使用WebRTC软件算法而使用平台内置算法。
src/media/engine/webrtcvoiceengine.cc
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: "
<< options_in.ToString();
AudioOptions options = options_in; // The options are modified below.
// Set and adjust echo canceller options.
// kEcConference is AEC with high suppression.
webrtc::EcModes ec_mode = webrtc::kEcConference;
#if defined(WEBRTC_IOS)
// wbt: 在ios上强制软件回声消除
if (options.ios_force_software_aec_HACK &&
*options.ios_force_software_aec_HACK) {
// EC may be forced on for a device known to have non-functioning platform
// AEC.
options.echo_cancellation = true;
options.extended_filter_aec = true;
RTC_LOG(LS_WARNING)
<< "Force software AEC on iOS. May conflict with platform AEC.";
} else {// wbt: 默认使用VPIO内置的回声消除
// On iOS, VPIO provides built-in EC.
options.echo_cancellation = false;
options.extended_filter_aec = false;
RTC_LOG(LS_INFO) << "Always disable AEC on iOS. Use built-in instead.";
}
#elif defined(WEBRTC_ANDROID)
ec_mode = webrtc::kEcAecm;
options.extended_filter_aec = false;
#endif
// wbt :除iOS平台外：如果启用了估计延时不确定性Delay Agnostic设置则自动开启aec
// Delay Agnostic AEC automatically turns on EC if not set except on iOS
// where the feature is not supported.
bool use_delay_agnostic_aec = false;
#if !defined(WEBRTC_IOS)
if (options.delay_agnostic_aec) {
use_delay_agnostic_aec = *options.delay_agnostic_aec;
if (use_delay_agnostic_aec) {
options.echo_cancellation = true;
options.extended_filter_aec = true;
ec_mode = webrtc::kEcConference;
}
}
#endif
// Set and adjust noise suppressor options.
#if defined(WEBRTC_IOS)
// On iOS, VPIO provides built-in NS.
// wbt: iOS平台使用VIPIO内置的降噪。
// 关闭键盘声音检测（因为触摸设备没有键盘)，并且关闭实验性降噪。
options.noise_suppression = false;
options.typing_detection = false;
options.experimental_ns = false;
RTC_LOG(LS_INFO) << "Always disable NS on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
// wbt: Android平台
// 关闭键盘声音检测（因为触摸设备没有键盘)，并且关闭实验性降噪。
options.typing_detection = false;
options.experimental_ns = false;
#endif
// Set and adjust gain control options.
#if defined(WEBRTC_IOS)
// On iOS, VPIO provides built-in AGC.
// wbt: iOS平台使用VIPIO内置的AGC增益。并且关闭试验性AGC.
options.auto_gain_control = false;
options.experimental_agc = false;
RTC_LOG(LS_INFO) << "Always disable AGC on iOS. Use built-in instead.";
#elif defined(WEBRTC_ANDROID)
// wbt: Android平台关闭试验性AGC.
options.experimental_agc = false;
#endif
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
// wbt: 手机平台
// 如果设置了"WebRTC-Audio-MinimizeResamplingOnMobile"则关闭AGC。
// 然后如果开启降噪且未开启回声消除则关闭高通滤波器。
// Turn off the gain control if specified by the field trial.
// The purpose of the field trial is to reduce the amount of resampling
// performed inside the audio processing module on mobile platforms by
// whenever possible turning off the fixed AGC mode and the high-pass filter.
// (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181).
if (webrtc::field_trial::IsEnabled(
"WebRTC-Audio-MinimizeResamplingOnMobile")) {
options.auto_gain_control = false;
RTC_LOG(LS_INFO) << "Disable AGC according to field trial.";
if (!(options.noise_suppression.value_or(false) ||
options.echo_cancellation.value_or(false))) {
// If possible, turn off the high-pass filter.
RTC_LOG(LS_INFO)
<< "Disable high-pass filter in response to field trial.";
options.highpass_filter = false;
}
}
#endif
if (options.echo_cancellation) {
// Check if platform supports built-in EC. Currently only supported on
// Android and in combination with Java based audio layer.
// TODO(henrika): investigate possibility to support built-in EC also
// in combination with Open SL ES audio.
// wbt: 目前只有android支持内置aec
// 如果支持内置aec而且开启回声消除echo_cancellation和未开启use_delay_agnostic_aec
// 且内置EnableBuiltInAEC时关闭软件回声消除从而使用平台内置回声消除。
const bool built_in_aec = adm()->BuiltInAECIsAvailable();//除了Android平台为true，其他平台默认为false
if (built_in_aec) {
// Built-in EC exists on this device and use_delay_agnostic_aec is not
// overriding it. Enable/Disable it according to the echo_cancellation
// audio option.
const bool enable_built_in_aec =
*options.echo_cancellation && !use_delay_agnostic_aec;
if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
enable_built_in_aec) {
// Disable internal software EC if built-in EC is enabled,
// i.e., replace the software EC with the built-in EC.
options.echo_cancellation = false;
RTC_LOG(LS_INFO)
<< "Disabling EC since built-in EC will be used instead";
}
}
// 这里的ec_mode在上面会根据情况设置为默认会议积极的aec还是手机型aec。
webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation,
ec_mode);
}
// 判断是否支持平台内置agc,如果支持则关闭软件agc
if (options.auto_gain_control) {
bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();
if (built_in_agc_avaliable) {
if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
*options.auto_gain_control) {
// Disable internal software AGC if built-in AGC is enabled,
// i.e., replace the software AGC with the built-in AGC.
options.auto_gain_control = false;
RTC_LOG(LS_INFO)
<< "Disabling AGC since built-in AGC will be used instead";
}
}
webrtc::apm_helpers::SetAgcStatus(apm(), *options.auto_gain_control);
}
//agc参数设置
if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
options.tx_agc_limiter) {
// Override default_agc_config_. Generally, an unset option means "leave
// the VoE bits alone" in this function, so we want whatever is set to be
// stored as the new "default". If we didn't, then setting e.g.
// tx_agc_target_dbov would reset digital compression gain and limiter
// settings.
default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or(
default_agc_config_.targetLeveldBOv);
default_agc_config_.digitalCompressionGaindB =
options.tx_agc_digital_compression_gain.value_or(
default_agc_config_.digitalCompressionGaindB);
default_agc_config_.limiterEnable =
options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);
webrtc::apm_helpers::SetAgcConfig(apm(), default_agc_config_);
}
// 如果支持平台内置降噪则关闭软件降噪
if (options.noise_suppression) {
if (adm()->BuiltInNSIsAvailable()) {
bool builtin_ns = *options.noise_suppression;
if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
// Disable internal software NS if built-in NS is enabled,
// i.e., replace the software NS with the built-in NS.
options.noise_suppression = false;
RTC_LOG(LS_INFO)
<< "Disabling NS since built-in NS will be used instead";
}
}
webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression);
}
// 立体声通道交换
if (options.stereo_swapping) {
RTC_LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
audio_state()->SetStereoChannelSwapping(*options.stereo_swapping);
}
// jitter buffer最大包数设置
if (options.audio_jitter_buffer_max_packets) {
RTC_LOG(LS_INFO) << "NetEq capacity is "
<< *options.audio_jitter_buffer_max_packets;
audio_jitter_buffer_max_packets_ =
std::max(20, *options.audio_jitter_buffer_max_packets);
}
// jitter buffer加速设置
if (options.audio_jitter_buffer_fast_accelerate) {
RTC_LOG(LS_INFO) << "NetEq fast mode? "
<< *options.audio_jitter_buffer_fast_accelerate;
audio_jitter_buffer_fast_accelerate_ =
*options.audio_jitter_buffer_fast_accelerate;
}
// jitter buffer最小延迟（毫秒）
if (options.audio_jitter_buffer_min_delay_ms) {
RTC_LOG(LS_INFO) << "NetEq minimum delay is "
<< *options.audio_jitter_buffer_min_delay_ms;
audio_jitter_buffer_min_delay_ms_ =
*options.audio_jitter_buffer_min_delay_ms;
}
// 键盘声音检测
if (options.typing_detection) {
RTC_LOG(LS_INFO) << "Typing detection is enabled? "
<< *options.typing_detection;
webrtc::apm_helpers::SetTypingDetectionStatus(apm(),
*options.typing_detection);
}
webrtc::Config config;
if (options.delay_agnostic_aec)
delay_agnostic_aec_ = options.delay_agnostic_aec;
if (delay_agnostic_aec_) {
RTC_LOG(LS_INFO) << "Delay agnostic aec is enabled? "
<< *delay_agnostic_aec_;
config.Set<webrtc::DelayAgnostic>(
new webrtc::DelayAgnostic(*delay_agnostic_aec_));
}
if (options.extended_filter_aec) {
extended_filter_aec_ = options.extended_filter_aec;
}
if (extended_filter_aec_) {
RTC_LOG(LS_INFO) << "Extended filter aec is enabled? "
<< *extended_filter_aec_;
config.Set<webrtc::ExtendedFilter>(
new webrtc::ExtendedFilter(*extended_filter_aec_));
}
if (options.experimental_ns) {
experimental_ns_ = options.experimental_ns;
}
if (experimental_ns_) {
RTC_LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
config.Set<webrtc::ExperimentalNs>(
new webrtc::ExperimentalNs(*experimental_ns_));
}
webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();
if (options.highpass_filter) {
apm_config.high_pass_filter.enabled = *options.highpass_filter;
}
// 残留回声检测
if (options.residual_echo_detector) {
apm_config.residual_echo_detector.enabled = *options.residual_echo_detector;
}
apm()->SetExtraOptions(config);
apm()->ApplyConfig(apm_config);
return true;
}
虽然没做android，但是顺便看了一下android上使用平台内置硬件AEC，NS的代码，代码在：
src/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
在java中使用AudioEffect检测是否支持硬件AEC和NS，使用AcousticEchoCanceler和NoiseSuppressor处理AEC和NS。
而AGC没有平台硬件支持，直接使用WebRTC中的算法。
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