/* * 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 "webrtc/modules/audio_processing/include/audio_processing.h" #include #include #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" namespace webrtc { class AgcManagerDirect; class AudioBuffer; 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) : rate_(sample_rate_hz), samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {} 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); // Only for testing. AudioProcessingImpl(const Config& config, Beamformer* 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; int InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_); int 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 gain_control_for_new_agc_; std::list component_list_; CriticalSectionWrapper* crit_; rtc::scoped_ptr render_audio_; rtc::scoped_ptr 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 debug_file_; rtc::scoped_ptr 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_; bool key_pressed_; // Only set through the constructor's Config parameter. const bool use_new_agc_; rtc::scoped_ptr agc_manager_ GUARDED_BY(crit_); bool transient_suppressor_enabled_; rtc::scoped_ptr transient_suppressor_; const bool beamformer_enabled_; rtc::scoped_ptr beamformer_; const std::vector array_geometry_; const bool supports_48kHz_; }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_