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/*
* Copyright (c) 2012 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_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#include <list>
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class AgcManagerDirect;
class AudioBuffer;
template<typename T>
class Beamformer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
class FileWrapper;
class GainControlImpl;
class GainControlForNewAgc;
class HighPassFilterImpl;
class LevelEstimatorImpl;
class NoiseSuppressionImpl;
class ProcessingComponent;
class TransientSuppressor;
class VoiceDetectionImpl;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
namespace audioproc {
class Event;
} // namespace audioproc
#endif
class AudioRate {
public:
explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
virtual ~AudioRate() {}
void set(int rate) {
rate_ = rate;
samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
}
int rate() const { return rate_; }
int samples_per_channel() const { return samples_per_channel_; }
private:
int rate_;
int samples_per_channel_;
};
class AudioFormat : public AudioRate {
public:
AudioFormat(int sample_rate_hz, int num_channels)
: AudioRate(sample_rate_hz),
num_channels_(num_channels) {}
virtual ~AudioFormat() {}
void set(int rate, int num_channels) {
AudioRate::set(rate);
num_channels_ = num_channels;
}
int num_channels() const { return num_channels_; }
private:
int num_channels_;
};
class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
// AudioProcessingImpl takes ownership of beamformer.
AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
virtual ~AudioProcessingImpl();
// AudioProcessing methods.
int Initialize() override;
int Initialize(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) override;
void SetExtraOptions(const Config& config) override;
int set_sample_rate_hz(int rate) override;
int input_sample_rate_hz() const override;
int sample_rate_hz() const override;
int proc_sample_rate_hz() const override;
int proc_split_sample_rate_hz() const override;
int num_input_channels() const override;
int num_output_channels() const override;
int num_reverse_channels() const override;
void set_output_will_be_muted(bool muted) override;
bool output_will_be_muted() const override;
int ProcessStream(AudioFrame* frame) override;
int ProcessStream(const float* const* src,
int samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest) override;
int AnalyzeReverseStream(AudioFrame* frame) override;
int AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) override;
int set_stream_delay_ms(int delay) override;
int stream_delay_ms() const override;
bool was_stream_delay_set() const override;
void set_delay_offset_ms(int offset) override;
int delay_offset_ms() const override;
void set_stream_key_pressed(bool key_pressed) override;
bool stream_key_pressed() const override;
int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
int StartDebugRecording(FILE* handle) override;
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
int StopDebugRecording() override;
EchoCancellation* echo_cancellation() const override;
EchoControlMobile* echo_control_mobile() const override;
GainControl* gain_control() const override;
HighPassFilter* high_pass_filter() const override;
LevelEstimator* level_estimator() const override;
NoiseSuppression* noise_suppression() const override;
VoiceDetection* voice_detection() const override;
protected:
// Overridden in a mock.
virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
private:
int InitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int MaybeInitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
bool is_data_processed() const;
bool output_copy_needed(bool is_data_processed) const;
bool synthesis_needed(bool is_data_processed) const;
bool analysis_needed(bool is_data_processed) const;
void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
GainControlImpl* gain_control_;
HighPassFilterImpl* high_pass_filter_;
LevelEstimatorImpl* level_estimator_;
NoiseSuppressionImpl* noise_suppression_;
VoiceDetectionImpl* voice_detection_;
rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
rtc::scoped_ptr<AudioBuffer> render_audio_;
rtc::scoped_ptr<AudioBuffer> capture_audio_;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
int WriteMessageToDebugFile();
int WriteInitMessage();
rtc::scoped_ptr<FileWrapper> debug_file_;
rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
std::string event_str_; // Memory for protobuf serialization.
#endif
AudioFormat fwd_in_format_;
// This one is an AudioRate, because the forward processing number of channels
// is mutable and is tracked by the capture_audio_.
AudioRate fwd_proc_format_;
AudioFormat fwd_out_format_;
AudioFormat rev_in_format_;
AudioFormat rev_proc_format_;
int split_rate_;
int stream_delay_ms_;
int delay_offset_ms_;
bool was_stream_delay_set_;
bool output_will_be_muted_ GUARDED_BY(crit_);
bool key_pressed_;
// Only set through the constructor's Config parameter.
const bool use_new_agc_;
rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
int agc_startup_min_volume_;
bool transient_suppressor_enabled_;
rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
const bool beamformer_enabled_;
rtc::scoped_ptr<Beamformer<float>> beamformer_;
const std::vector<Point> array_geometry_;
const bool supports_48kHz_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_