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helpers.cc
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// Copyright 2019 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/webrtc/helpers.h"
#include "base/feature_list.h"
#include "base/files/file_util.h"
#include "base/logging.h"
#include "build/build_config.h"
#include "build/chromecast_buildflags.h"
#include "media/webrtc/webrtc_features.h"
#include "third_party/webrtc/api/audio/echo_canceller3_config.h"
#include "third_party/webrtc/api/audio/echo_canceller3_factory.h"
#include "third_party/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
namespace media {
namespace {
// The analog gain controller is not supported on mobile - i.e., Android, iOS.
#if defined(OS_ANDROID) || defined(OS_IOS)
constexpr bool kAnalogAgcSupported = false;
#else
constexpr bool kAnalogAgcSupported = true;
#endif // defined(OS_ANDROID) || defined(OS_IOS)
// The analog gain controller can only be disabled on Chromecast.
#if BUILDFLAG(IS_CHROMECAST)
constexpr bool kAllowToDisableAnalogAgc = true;
#else
constexpr bool kAllowToDisableAnalogAgc = false;
#endif // BUILDFLAG(IS_CHROMECAST)
// AGC1 mode.
using Agc1Mode = webrtc::AudioProcessing::Config::GainController1::Mode;
// TODO(bugs.webrtc.org/7909): Maybe set mode to kFixedDigital also for IOS.
#if defined(OS_ANDROID)
constexpr Agc1Mode kAgc1Mode = Agc1Mode::kFixedDigital;
#else
constexpr Agc1Mode kAgc1Mode = Agc1Mode::kAdaptiveAnalog;
#endif
using Agc1AnalagConfig =
::webrtc::AudioProcessing::Config::GainController1::AnalogGainController;
Agc1AnalagConfig::ClippingPredictor::Mode GetClippingPredictorMode(int mode) {
using Mode = Agc1AnalagConfig::ClippingPredictor::Mode;
switch (mode) {
case 1:
return Mode::kAdaptiveStepClippingPeakPrediction;
case 2:
return Mode::kFixedStepClippingPeakPrediction;
default:
return Mode::kClippingEventPrediction;
}
}
bool Allow48kHzApmProcessing() {
return base::FeatureList::IsEnabled(
::features::kWebRtcAllow48kHzProcessingOnArm);
}
absl::optional<int> GetAgcStartupMinVolume() {
if (!base::FeatureList::IsEnabled(
::features::kWebRtcAnalogAgcStartupMinVolume)) {
return absl::nullopt;
}
return base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcStartupMinVolume, "volume", 0);
}
void ConfigAgc2AdaptiveDigitalForHybridExperiment(
::webrtc::AudioProcessing::Config::GainController2::AdaptiveDigital&
config) {
config.dry_run = base::GetFieldTrialParamByFeatureAsBool(
::features::kWebRtcHybridAgc, "dry_run", false);
config.vad_reset_period_ms = base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcHybridAgc, "vad_reset_period_ms", 1500);
config.adjacent_speech_frames_threshold =
base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcHybridAgc, "adjacent_speech_frames_threshold", 12);
config.max_gain_change_db_per_second =
static_cast<float>(base::GetFieldTrialParamByFeatureAsDouble(
::features::kWebRtcHybridAgc, "max_gain_change_db_per_second", 3));
config.max_output_noise_level_dbfs =
static_cast<float>(base::GetFieldTrialParamByFeatureAsDouble(
::features::kWebRtcHybridAgc, "max_output_noise_level_dbfs", -50));
}
void ConfigAgc1AnalogForClippingControlExperiment(Agc1AnalagConfig& config) {
config.clipped_level_step = base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl, "clipped_level_step", 15);
config.clipped_ratio_threshold =
static_cast<float>(base::GetFieldTrialParamByFeatureAsDouble(
::features::kWebRtcAnalogAgcClippingControl,
"clipped_ratio_threshold", 0.1));
config.clipped_wait_frames = base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl, "clipped_wait_frames", 300);
config.clipping_predictor.mode =
GetClippingPredictorMode(base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl, "mode", 0));
config.clipping_predictor.window_length =
base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl, "window_length", 5);
config.clipping_predictor.reference_window_length =
base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl,
"reference_window_length", 5);
config.clipping_predictor.reference_window_delay =
base::GetFieldTrialParamByFeatureAsInt(
::features::kWebRtcAnalogAgcClippingControl, "reference_window_delay",
5);
config.clipping_predictor.clipping_threshold =
static_cast<float>(base::GetFieldTrialParamByFeatureAsDouble(
::features::kWebRtcAnalogAgcClippingControl, "clipping_threshold",
-1.0));
config.clipping_predictor.crest_factor_margin =
static_cast<float>(base::GetFieldTrialParamByFeatureAsDouble(
::features::kWebRtcAnalogAgcClippingControl, "crest_factor_margin",
3.0));
config.clipping_predictor.use_predicted_step =
base::GetFieldTrialParamByFeatureAsBool(
::features::kWebRtcAnalogAgcClippingControl, "use_predicted_step",
true);
}
// Configures automatic gain control in `apm_config`.
// TODO(bugs.webrtc.org/7494): Clean up once hybrid AGC experiment finalized.
// TODO(bugs.webrtc.org/7494): Remove unused cases, simplify decision logic.
void ConfigAutomaticGainControl(const AudioProcessingSettings& settings,
webrtc::AudioProcessing::Config& apm_config) {
// Configure AGC1.
if (settings.automatic_gain_control) {
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode = kAgc1Mode;
}
auto& agc1_analog_config = apm_config.gain_controller1.analog_gain_controller;
// Enable and configure AGC1 Analog if needed.
if (kAnalogAgcSupported && settings.experimental_automatic_gain_control) {
agc1_analog_config.enabled = true;
absl::optional<int> startup_min_volume = GetAgcStartupMinVolume();
// TODO(crbug.com/555577): Do not zero if `startup_min_volume` if no
// override is specified, instead fall back to the config default value.
agc1_analog_config.startup_min_volume = startup_min_volume.value_or(0);
}
// Disable AGC1 Analog.
if (kAllowToDisableAnalogAgc &&
!settings.experimental_automatic_gain_control) {
// This should likely be done on non-Chromecast platforms as well, but care
// is needed since users may be relying on the current behavior.
// https://crbug.com/918677#c4
agc1_analog_config.enabled = false;
}
// TODO(bugs.webrtc.org/7909): Consider returning if `kAnalogAgcSupported` is
// false since the AGC clipping controller and the Hybrid AGC experiments are
// meant to run when AGC1 Analog is used.
if (!settings.automatic_gain_control ||
!settings.experimental_automatic_gain_control ||
!agc1_analog_config.enabled) {
// The settings below only apply when AGC is enabled and when the analog
// controller is supported and enabled.
return;
}
// AGC1 Analog Clipping Controller experiment.
if (base::FeatureList::IsEnabled(
::features::kWebRtcAnalogAgcClippingControl)) {
agc1_analog_config.clipping_predictor.enabled = true;
ConfigAgc1AnalogForClippingControlExperiment(agc1_analog_config);
}
// Hybrid AGC feature.
const bool use_hybrid_agc =
base::FeatureList::IsEnabled(::features::kWebRtcHybridAgc);
auto& agc2_config = apm_config.gain_controller2;
agc2_config.enabled = use_hybrid_agc;
agc2_config.fixed_digital.gain_db = 0.0f;
if (use_hybrid_agc) {
agc2_config.adaptive_digital.enabled = true;
ConfigAgc2AdaptiveDigitalForHybridExperiment(agc2_config.adaptive_digital);
// Disable AGC1 adaptive digital unless AGC2 adaptive digital runs in
// dry-run mode.
agc1_analog_config.enable_digital_adaptive =
agc2_config.adaptive_digital.dry_run;
} else {
// Use the adaptive digital controller of AGC1 and disable that of AGC2.
agc1_analog_config.enable_digital_adaptive = true;
agc2_config.adaptive_digital.enabled = false;
}
}
} // namespace
webrtc::StreamConfig CreateStreamConfig(const AudioParameters& parameters) {
int channels = parameters.channels();
// Mapping all discrete channel layouts to max two channels assuming that any
// required channel remix takes place in the native audio layer.
if (parameters.channel_layout() == CHANNEL_LAYOUT_DISCRETE) {
channels = std::min(parameters.channels(), 2);
}
const int rate = parameters.sample_rate();
const bool has_keyboard =
parameters.channel_layout() == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC;
// webrtc::StreamConfig requires that the keyboard mic channel is not included
// in the channel count. It may still be used.
if (has_keyboard)
channels -= 1;
return webrtc::StreamConfig(rate, channels, has_keyboard);
}
bool LeftAndRightChannelsAreSymmetric(const AudioBus& audio) {
if (audio.channels() <= 1) {
return true;
}
return std::equal(audio.channel(0), audio.channel(0) + audio.frames(),
audio.channel(1));
}
void StartEchoCancellationDump(webrtc::AudioProcessing* audio_processing,
base::File aec_dump_file,
rtc::TaskQueue* worker_queue) {
DCHECK(aec_dump_file.IsValid());
FILE* stream = base::FileToFILE(std::move(aec_dump_file), "w");
if (!stream) {
LOG(DFATAL) << "Failed to open AEC dump file";
return;
}
auto aec_dump = webrtc::AecDumpFactory::Create(
stream, -1 /* max_log_size_bytes */, worker_queue);
if (!aec_dump) {
LOG(ERROR) << "Failed to start AEC debug recording";
return;
}
audio_processing->AttachAecDump(std::move(aec_dump));
}
void StopEchoCancellationDump(webrtc::AudioProcessing* audio_processing) {
audio_processing->DetachAecDump();
}
rtc::scoped_refptr<webrtc::AudioProcessing> CreateWebRtcAudioProcessingModule(
const AudioProcessingSettings& settings) {
// Create and configure the webrtc::AudioProcessing.
webrtc::AudioProcessingBuilder ap_builder;
if (settings.echo_cancellation) {
ap_builder.SetEchoControlFactory(
std::make_unique<webrtc::EchoCanceller3Factory>());
}
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing_module =
ap_builder.Create();
webrtc::AudioProcessing::Config apm_config =
audio_processing_module->GetConfig();
apm_config.pipeline.multi_channel_render = true;
apm_config.pipeline.multi_channel_capture =
settings.multi_channel_capture_processing;
apm_config.high_pass_filter.enabled = settings.high_pass_filter;
apm_config.noise_suppression.enabled = settings.noise_suppression;
apm_config.noise_suppression.level =
webrtc::AudioProcessing::Config::NoiseSuppression::Level::kHigh;
apm_config.echo_canceller.enabled = settings.echo_cancellation;
#if defined(OS_ANDROID)
apm_config.echo_canceller.mobile_mode = true;
#else
apm_config.echo_canceller.mobile_mode = false;
#endif
apm_config.residual_echo_detector.enabled = false;
#if !(defined(OS_ANDROID) || defined(OS_IOS))
apm_config.transient_suppression.enabled =
settings.transient_noise_suppression;
#endif
ConfigAutomaticGainControl(settings, apm_config);
// Ensure that 48 kHz APM processing is always active. This overrules the
// default setting in WebRTC of 32 kHz for ARM platforms.
if (Allow48kHzApmProcessing()) {
apm_config.pipeline.maximum_internal_processing_rate = 48000;
}
audio_processing_module->ApplyConfig(apm_config);
return audio_processing_module;
}
} // namespace media