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// Copyright 2013 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.
#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_MEDIASTREAM_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_MEDIASTREAM_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
#include <string>
#include "base/files/file.h"
#include "base/macros.h"
#include "base/optional.h"
#include "base/threading/thread_checker.h"
#include "build/build_config.h"
#include "media/base/audio_point.h"
#include "media/base/audio_processing.h"
#include "third_party/blink/public/common/mediastream/media_stream_request.h"
#include "third_party/blink/renderer/platform/mediastream/media_constraints.h"
#include "third_party/blink/renderer/platform/platform_export.h"
#include "third_party/webrtc/api/media_stream_interface.h"
#include "third_party/webrtc/media/base/media_channel.h"
#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
#include "third_party/webrtc/rtc_base/task_queue.h"
namespace webrtc {
class TypingDetection;
}
namespace blink {
using webrtc::AudioProcessing;
static constexpr int kAudioProcessingSampleRate =
#if defined(OS_ANDROID)
AudioProcessing::kSampleRate16kHz;
#else
AudioProcessing::kSampleRate48kHz;
#endif
// Simple struct with audio-processing properties.
struct PLATFORM_EXPORT AudioProcessingProperties {
enum class EchoCancellationType {
// Echo cancellation disabled.
kEchoCancellationDisabled,
// The WebRTC-provided AEC3 echo canceller.
kEchoCancellationAec3,
// System echo canceller, for example an OS-provided or hardware echo
// canceller.
kEchoCancellationSystem
};
// Creates an AudioProcessingProperties object with fields initialized to
// their default values.
AudioProcessingProperties();
AudioProcessingProperties(const AudioProcessingProperties& other);
AudioProcessingProperties& operator=(const AudioProcessingProperties& other);
// Disables properties that are enabled by default.
void DisableDefaultProperties();
// Returns whether echo cancellation is enabled.
bool EchoCancellationEnabled() const;
// Returns whether WebRTC-provided echo cancellation is enabled.
bool EchoCancellationIsWebRtcProvided() const;
bool HasSameReconfigurableSettings(
const AudioProcessingProperties& other) const;
bool HasSameNonReconfigurableSettings(
const AudioProcessingProperties& other) const;
// Converts this struct to an equivalent media::AudioProcessingSettings.
// TODO(https://crbug.com/878757): Eliminate this class in favor of the media
// one.
media::AudioProcessingSettings ToAudioProcessingSettings() const;
EchoCancellationType echo_cancellation_type =
EchoCancellationType::kEchoCancellationAec3;
bool disable_hw_noise_suppression = false;
bool goog_audio_mirroring = false;
bool goog_auto_gain_control = true;
bool goog_experimental_echo_cancellation =
#if defined(OS_ANDROID)
false;
#else
true;
#endif
bool goog_typing_noise_detection = false;
bool goog_noise_suppression = true;
bool goog_experimental_noise_suppression = true;
bool goog_highpass_filter = true;
bool goog_experimental_auto_gain_control = true;
};
// Enables the typing detection with the given detector.
PLATFORM_EXPORT void EnableTypingDetection(
AudioProcessing::Config* apm_config,
webrtc::TypingDetection* typing_detector);
// Starts the echo cancellation dump in
// |audio_processing|. |worker_queue| must be kept alive until either
// |audio_processing| is destroyed, or
// StopEchoCancellationDump(audio_processing) is called.
PLATFORM_EXPORT void StartEchoCancellationDump(
AudioProcessing* audio_processing,
base::File aec_dump_file,
rtc::TaskQueue* worker_queue);
// Stops the echo cancellation dump in |audio_processing|.
// This method has no impact if echo cancellation dump has not been started on
// |audio_processing|.
PLATFORM_EXPORT void StopEchoCancellationDump(
AudioProcessing* audio_processing);
// Adaptive Gain Controller 2 (aka AGC2) properties.
struct PLATFORM_EXPORT AdaptiveGainController2Properties {
float vad_probability_attack;
bool use_peaks_not_rms;
int level_estimator_speech_frames_threshold;
int initial_saturation_margin_db;
int extra_saturation_margin_db;
int gain_applier_speech_frames_threshold;
int max_gain_change_db_per_second;
int max_output_noise_level_dbfs;
};
// Configures automatic gain control in `apm_config`. If `agc_enabled` is true
// and `agc2_properties` is specified, the AGC2 adaptive digital replaces the
// adaptive digital controller of AGC1 - i.e., hybrid configuration (AGC1 analog
// plus AGC2 adaptive digital).
// TODO(crbug.com/webrtc/7494): Clean up once hybrid AGC experiment finalized.
PLATFORM_EXPORT void ConfigAutomaticGainControl(
bool agc_enabled,
bool experimental_agc_enabled,
base::Optional<AdaptiveGainController2Properties> agc2_properties,
base::Optional<double> compression_gain_db,
AudioProcessing::Config& apm_config);
PLATFORM_EXPORT void PopulateApmConfig(
AudioProcessing::Config* apm_config,
const AudioProcessingProperties& properties,
const base::Optional<std::string>& audio_processing_platform_config_json,
base::Optional<double>* gain_control_compression_gain_db);
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_PLATFORM_MEDIASTREAM_MEDIA_STREAM_AUDIO_PROCESSOR_OPTIONS_H_
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