blob: 84bba85939419b82886e882c75005b791ef08926 [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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/rtc_base/array_view.h"
namespace webrtc {
class AudioFrame;
// Struct for passing current config from APM without having to
// include protobuf headers.
struct InternalAPMConfig {
InternalAPMConfig();
InternalAPMConfig(const InternalAPMConfig&);
InternalAPMConfig(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 intelligibility_enhancer_enabled = false;
bool noise_robust_agc_enabled = false;
std::string experiments_description = "";
};
struct InternalAPMStreamsConfig {
int input_sample_rate = 0;
int output_sample_rate = 0;
int render_input_sample_rate = 0;
int render_output_sample_rate = 0;
size_t input_num_channels = 0;
size_t output_num_channels = 0;
size_t render_input_num_channels = 0;
size_t render_output_num_channels = 0;
};
// Class to pass audio data in float** format. This is to avoid
// dependence on AudioBuffer, and avoid problems associated with
// rtc::ArrayView<rtc::ArrayView>.
class FloatAudioFrame {
public:
// |num_channels| and |channel_size| describe the float**
// |audio_samples|. |audio_samples| is assumed to point to a
// two-dimensional |num_channels * channel_size| array of floats.
FloatAudioFrame(const float* const* audio_samples,
size_t num_channels,
size_t channel_size)
: audio_samples_(audio_samples),
num_channels_(num_channels),
channel_size_(channel_size) {}
FloatAudioFrame() = delete;
size_t num_channels() const { return num_channels_; }
rtc::ArrayView<const float> channel(size_t idx) const {
RTC_DCHECK_LE(0, idx);
RTC_DCHECK_LE(idx, num_channels_);
return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_);
}
private:
const float* const* audio_samples_;
size_t num_channels_;
size_t channel_size_;
};
// 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 {
public:
struct AudioProcessingState {
int delay;
int drift;
int level;
bool keypress;
};
virtual ~AecDump() = default;
// Logs Event::Type INIT message.
virtual void WriteInitMessage(
const InternalAPMStreamsConfig& streams_config) = 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 FloatAudioFrame& src) = 0;
virtual void AddCaptureStreamOutput(const FloatAudioFrame& 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 FloatAudioFrame& src) = 0;
// Logs Event::Type CONFIG message.
virtual void WriteConfig(const InternalAPMConfig& config) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_