blob: b734adfbbbb36a777e3ec9b6bb99f4505fdcf20f [file] [log] [blame]
* Copyright (c) 2017 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.
#include <stdint.h>
#include <string>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/deprecation.h"
namespace webrtc {
// Struct for passing current config from APM without having to
// include protobuf headers.
struct InternalAPMConfig {
InternalAPMConfig(const InternalAPMConfig&);
InternalAPMConfig& operator=(const InternalAPMConfig&);
InternalAPMConfig& operator=(InternalAPMConfig&&) = delete;
bool operator==(const InternalAPMConfig& other);
bool aec_enabled = false;
bool aec_delay_agnostic_enabled = false;
bool aec_drift_compensation_enabled = false;
bool aec_extended_filter_enabled = false;
int aec_suppression_level = 0;
bool aecm_enabled = false;
bool aecm_comfort_noise_enabled = false;
int aecm_routing_mode = 0;
bool agc_enabled = false;
int agc_mode = 0;
bool agc_limiter_enabled = false;
bool hpf_enabled = false;
bool ns_enabled = false;
int ns_level = 0;
bool transient_suppression_enabled = false;
bool noise_robust_agc_enabled = false;
bool pre_amplifier_enabled = false;
float pre_amplifier_fixed_gain_factor = 1.f;
std::string experiments_description = "";
// An interface for recording configuration and input/output streams
// of the Audio Processing Module. The recordings are called
// 'aec-dumps' and are stored in a protobuf format defined in
// debug.proto.
// The Write* methods are always safe to call concurrently or
// otherwise for all implementing subclasses. The intended mode of
// operation is to create a protobuf object from the input, and send
// it away to be written to file asynchronously.
class AecDump {
struct AudioProcessingState {
int delay;
int drift;
int level;
bool keypress;
virtual ~AecDump() = default;
// Logs Event::Type INIT message.
virtual void WriteInitMessage(const ProcessingConfig& api_format,
int64_t time_now_ms) = 0;
RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
WriteInitMessage(api_format, 0);
// Logs Event::Type STREAM message. To log an input/output pair,
// call the AddCapture* and AddAudioProcessingState methods followed
// by a WriteCaptureStreamMessage call.
virtual void AddCaptureStreamInput(
const AudioFrameView<const float>& src) = 0;
virtual void AddCaptureStreamOutput(
const AudioFrameView<const float>& src) = 0;
virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0;
virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0;
virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
virtual void WriteCaptureStreamMessage() = 0;
// Logs Event::Type REVERSE_STREAM message.
virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
virtual void WriteRenderStreamMessage(
const AudioFrameView<const float>& src) = 0;
virtual void WriteRuntimeSetting(
const AudioProcessing::RuntimeSetting& runtime_setting) = 0;
// Logs Event::Type CONFIG message.
virtual void WriteConfig(const InternalAPMConfig& config) = 0;
} // namespace webrtc