| /* |
| * 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 MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
| #define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
| |
| #include <list> |
| #include <memory> |
| #include <vector> |
| |
| #include "api/function_view.h" |
| #include "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/include/aec_dump.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/include/audio_processing_statistics.h" |
| #include "modules/audio_processing/render_queue_item_verifier.h" |
| #include "modules/audio_processing/rms_level.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/gtest_prod_util.h" |
| #include "rtc_base/ignore_wundef.h" |
| #include "rtc_base/swap_queue.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| class ApmDataDumper; |
| class AudioConverter; |
| |
| class AudioProcessingImpl : public AudioProcessing { |
| public: |
| // Methods forcing APM to run in a single-threaded manner. |
| // Acquires both the render and capture locks. |
| explicit AudioProcessingImpl(const webrtc::Config& config); |
| // AudioProcessingImpl takes ownership of capture post processor. |
| AudioProcessingImpl(const webrtc::Config& config, |
| std::unique_ptr<CustomProcessing> capture_post_processor, |
| std::unique_ptr<CustomProcessing> render_pre_processor, |
| std::unique_ptr<EchoControlFactory> echo_control_factory, |
| rtc::scoped_refptr<EchoDetector> echo_detector, |
| std::unique_ptr<CustomAudioAnalyzer> capture_analyzer); |
| ~AudioProcessingImpl() override; |
| int Initialize() override; |
| int Initialize(int capture_input_sample_rate_hz, |
| int capture_output_sample_rate_hz, |
| int render_sample_rate_hz, |
| ChannelLayout capture_input_layout, |
| ChannelLayout capture_output_layout, |
| ChannelLayout render_input_layout) override; |
| int Initialize(const ProcessingConfig& processing_config) override; |
| void ApplyConfig(const AudioProcessing::Config& config) override; |
| void SetExtraOptions(const webrtc::Config& config) override; |
| void UpdateHistogramsOnCallEnd() override; |
| void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override; |
| void DetachAecDump() override; |
| void AttachPlayoutAudioGenerator( |
| std::unique_ptr<AudioGenerator> audio_generator) override; |
| void DetachPlayoutAudioGenerator() override; |
| |
| void SetRuntimeSetting(RuntimeSetting setting) override; |
| |
| // Capture-side exclusive methods possibly running APM in a |
| // multi-threaded manner. Acquire the capture lock. |
| int ProcessStream(AudioFrame* frame) override; |
| int ProcessStream(const float* const* src, |
| size_t samples_per_channel, |
| int input_sample_rate_hz, |
| ChannelLayout input_layout, |
| int output_sample_rate_hz, |
| ChannelLayout output_layout, |
| float* const* dest) override; |
| int ProcessStream(const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| float* const* dest) override; |
| void set_output_will_be_muted(bool muted) override; |
| int set_stream_delay_ms(int delay) override; |
| void set_delay_offset_ms(int offset) override; |
| int delay_offset_ms() const override; |
| void set_stream_key_pressed(bool key_pressed) override; |
| void set_stream_analog_level(int level) override; |
| int recommended_stream_analog_level() const override; |
| |
| // Render-side exclusive methods possibly running APM in a |
| // multi-threaded manner. Acquire the render lock. |
| int ProcessReverseStream(AudioFrame* frame) override; |
| int AnalyzeReverseStream(const float* const* data, |
| size_t samples_per_channel, |
| int sample_rate_hz, |
| ChannelLayout layout) override; |
| int ProcessReverseStream(const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config, |
| float* const* dest) override; |
| |
| // Methods only accessed from APM submodules or |
| // from AudioProcessing tests in a single-threaded manner. |
| // Hence there is no need for locks in these. |
| int proc_sample_rate_hz() const override; |
| int proc_split_sample_rate_hz() const override; |
| size_t num_input_channels() const override; |
| size_t num_proc_channels() const override; |
| size_t num_output_channels() const override; |
| size_t num_reverse_channels() const override; |
| int stream_delay_ms() const override; |
| bool was_stream_delay_set() const override |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| AudioProcessingStats GetStatistics(bool has_remote_tracks) const override; |
| |
| // Methods returning pointers to APM submodules. |
| // No locks are aquired in those, as those locks |
| // would offer no protection (the submodules are |
| // created only once in a single-treaded manner |
| // during APM creation). |
| GainControl* gain_control() const override; |
| LevelEstimator* level_estimator() const override; |
| NoiseSuppression* noise_suppression() const override; |
| |
| // TODO(peah): Remove MutateConfig once the new API allows that. |
| void MutateConfig(rtc::FunctionView<void(AudioProcessing::Config*)> mutator); |
| AudioProcessing::Config GetConfig() const override; |
| |
| protected: |
| // Overridden in a mock. |
| virtual int InitializeLocked() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| |
| private: |
| // TODO(peah): These friend classes should be removed as soon as the new |
| // parameter setting scheme allows. |
| FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior); |
| FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior); |
| FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior); |
| |
| // Class providing thread-safe message pipe functionality for |
| // |runtime_settings_|. |
| class RuntimeSettingEnqueuer { |
| public: |
| explicit RuntimeSettingEnqueuer( |
| SwapQueue<RuntimeSetting>* runtime_settings); |
| ~RuntimeSettingEnqueuer(); |
| void Enqueue(RuntimeSetting setting); |
| |
| private: |
| SwapQueue<RuntimeSetting>& runtime_settings_; |
| }; |
| struct ApmPublicSubmodules; |
| struct ApmPrivateSubmodules; |
| |
| std::unique_ptr<ApmDataDumper> data_dumper_; |
| static int instance_count_; |
| |
| SwapQueue<RuntimeSetting> capture_runtime_settings_; |
| SwapQueue<RuntimeSetting> render_runtime_settings_; |
| |
| RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_; |
| RuntimeSettingEnqueuer render_runtime_settings_enqueuer_; |
| |
| // EchoControl factory. |
| std::unique_ptr<EchoControlFactory> echo_control_factory_; |
| |
| class ApmSubmoduleStates { |
| public: |
| ApmSubmoduleStates(bool capture_post_processor_enabled, |
| bool render_pre_processor_enabled, |
| bool capture_analyzer_enabled); |
| // Updates the submodule state and returns true if it has changed. |
| bool Update(bool high_pass_filter_enabled, |
| bool echo_canceller_enabled, |
| bool mobile_echo_controller_enabled, |
| bool residual_echo_detector_enabled, |
| bool noise_suppressor_enabled, |
| bool adaptive_gain_controller_enabled, |
| bool gain_controller2_enabled, |
| bool pre_amplifier_enabled, |
| bool echo_controller_enabled, |
| bool voice_detector_enabled, |
| bool level_estimator_enabled, |
| bool transient_suppressor_enabled); |
| bool CaptureMultiBandSubModulesActive() const; |
| bool CaptureMultiBandProcessingPresent() const; |
| bool CaptureMultiBandProcessingActive(bool ec_processing_active) const; |
| bool CaptureFullBandProcessingActive() const; |
| bool CaptureAnalyzerActive() const; |
| bool RenderMultiBandSubModulesActive() const; |
| bool RenderFullBandProcessingActive() const; |
| bool RenderMultiBandProcessingActive() const; |
| bool HighPassFilteringRequired() const; |
| |
| private: |
| const bool capture_post_processor_enabled_ = false; |
| const bool render_pre_processor_enabled_ = false; |
| const bool capture_analyzer_enabled_ = false; |
| bool high_pass_filter_enabled_ = false; |
| bool echo_canceller_enabled_ = false; |
| bool mobile_echo_controller_enabled_ = false; |
| bool residual_echo_detector_enabled_ = false; |
| bool noise_suppressor_enabled_ = false; |
| bool adaptive_gain_controller_enabled_ = false; |
| bool gain_controller2_enabled_ = false; |
| bool pre_amplifier_enabled_ = false; |
| bool echo_controller_enabled_ = false; |
| bool level_estimator_enabled_ = false; |
| bool voice_detector_enabled_ = false; |
| bool transient_suppressor_enabled_ = false; |
| bool first_update_ = true; |
| }; |
| |
| // Method for modifying the formats struct that are called from both |
| // the render and capture threads. The check for whether modifications |
| // are needed is done while holding the render lock only, thereby avoiding |
| // that the capture thread blocks the render thread. |
| // The struct is modified in a single-threaded manner by holding both the |
| // render and capture locks. |
| int MaybeInitializeRender(const ProcessingConfig& processing_config) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Method for updating the state keeping track of the active submodules. |
| // Returns a bool indicating whether the state has changed. |
| bool UpdateActiveSubmoduleStates() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Methods requiring APM running in a single-threaded manner. |
| // Are called with both the render and capture locks already |
| // acquired. |
| void InitializeTransient() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| int InitializeLocked(const ProcessingConfig& config) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeResidualEchoDetector() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeHighPassFilter() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeVoiceDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeEchoController() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializePreAmplifier() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeAnalyzer() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Sample rate used for the fullband processing. |
| int proc_fullband_sample_rate_hz() const |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Empties and handles the respective RuntimeSetting queues. |
| void HandleCaptureRuntimeSettings() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void HandleRenderRuntimeSettings() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| void ApplyAgc1Config(const Config::GainController1& agc_config) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Returns a direct pointer to the AGC1 submodule: either a GainControlImpl |
| // or GainControlForExperimentalAgc instance. |
| GainControl* agc1(); |
| const GainControl* agc1() const; |
| |
| void EmptyQueuedRenderAudio(); |
| void AllocateRenderQueue() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void QueueBandedRenderAudio(AudioBuffer* audio) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| void QueueNonbandedRenderAudio(AudioBuffer* audio) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Capture-side exclusive methods possibly running APM in a multi-threaded |
| // manner that are called with the render lock already acquired. |
| int ProcessCaptureStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Render-side exclusive methods possibly running APM in a multi-threaded |
| // manner that are called with the render lock already acquired. |
| // TODO(ekm): Remove once all clients updated to new interface. |
| int AnalyzeReverseStreamLocked(const float* const* src, |
| const StreamConfig& input_config, |
| const StreamConfig& output_config) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| int ProcessRenderStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Collects configuration settings from public and private |
| // submodules to be saved as an audioproc::Config message on the |
| // AecDump if it is attached. If not |forced|, only writes the current |
| // config if it is different from the last saved one; if |forced|, |
| // writes the config regardless of the last saved. |
| void WriteAecDumpConfigMessage(bool forced) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Notifies attached AecDump of current configuration and capture data. |
| void RecordUnprocessedCaptureStream(const float* const* capture_stream) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| void RecordUnprocessedCaptureStream(const AudioFrame& capture_frame) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Notifies attached AecDump of current configuration and |
| // processed capture data and issues a capture stream recording |
| // request. |
| void RecordProcessedCaptureStream( |
| const float* const* processed_capture_stream) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| void RecordProcessedCaptureStream(const AudioFrame& processed_capture_frame) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Notifies attached AecDump about current state (delay, drift, etc). |
| void RecordAudioProcessingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // AecDump instance used for optionally logging APM config, input |
| // and output to file in the AEC-dump format defined in debug.proto. |
| std::unique_ptr<AecDump> aec_dump_; |
| |
| // Hold the last config written with AecDump for avoiding writing |
| // the same config twice. |
| InternalAPMConfig apm_config_for_aec_dump_ RTC_GUARDED_BY(crit_capture_); |
| |
| // Critical sections. |
| rtc::CriticalSection crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_); |
| rtc::CriticalSection crit_capture_; |
| |
| // Struct containing the Config specifying the behavior of APM. |
| AudioProcessing::Config config_; |
| |
| // Class containing information about what submodules are active. |
| ApmSubmoduleStates submodule_states_; |
| |
| // Structs containing the pointers to the submodules. |
| std::unique_ptr<ApmPublicSubmodules> public_submodules_; |
| std::unique_ptr<ApmPrivateSubmodules> private_submodules_; |
| |
| // State that is written to while holding both the render and capture locks |
| // but can be read without any lock being held. |
| // As this is only accessed internally of APM, and all internal methods in APM |
| // either are holding the render or capture locks, this construct is safe as |
| // it is not possible to read the variables while writing them. |
| struct ApmFormatState { |
| ApmFormatState() |
| : // Format of processing streams at input/output call sites. |
| api_format({{{kSampleRate16kHz, 1, false}, |
| {kSampleRate16kHz, 1, false}, |
| {kSampleRate16kHz, 1, false}, |
| {kSampleRate16kHz, 1, false}}}), |
| render_processing_format(kSampleRate16kHz, 1) {} |
| ProcessingConfig api_format; |
| StreamConfig render_processing_format; |
| } formats_; |
| |
| // APM constants. |
| const struct ApmConstants { |
| ApmConstants(int agc_startup_min_volume, |
| int agc_clipped_level_min, |
| bool use_experimental_agc, |
| bool use_experimental_agc_agc2_level_estimation, |
| bool use_experimental_agc_agc2_digital_adaptive, |
| bool use_experimental_agc_process_before_aec, |
| bool experimental_multi_channel_render_support, |
| bool experimental_multi_channel_capture_support) |
| : agc_startup_min_volume(agc_startup_min_volume), |
| agc_clipped_level_min(agc_clipped_level_min), |
| use_experimental_agc(use_experimental_agc), |
| use_experimental_agc_agc2_level_estimation( |
| use_experimental_agc_agc2_level_estimation), |
| use_experimental_agc_agc2_digital_adaptive( |
| use_experimental_agc_agc2_digital_adaptive), |
| use_experimental_agc_process_before_aec( |
| use_experimental_agc_process_before_aec), |
| experimental_multi_channel_render_support( |
| experimental_multi_channel_render_support), |
| experimental_multi_channel_capture_support( |
| experimental_multi_channel_capture_support) {} |
| int agc_startup_min_volume; |
| int agc_clipped_level_min; |
| bool use_experimental_agc; |
| bool use_experimental_agc_agc2_level_estimation; |
| bool use_experimental_agc_agc2_digital_adaptive; |
| bool use_experimental_agc_process_before_aec; |
| bool experimental_multi_channel_render_support; |
| bool experimental_multi_channel_capture_support; |
| } constants_; |
| |
| struct ApmCaptureState { |
| ApmCaptureState(bool transient_suppressor_enabled); |
| ~ApmCaptureState(); |
| int delay_offset_ms; |
| bool was_stream_delay_set; |
| bool output_will_be_muted; |
| bool key_pressed; |
| bool transient_suppressor_enabled; |
| std::unique_ptr<AudioBuffer> capture_audio; |
| std::unique_ptr<AudioBuffer> capture_fullband_audio; |
| // Only the rate and samples fields of capture_processing_format_ are used |
| // because the capture processing number of channels is mutable and is |
| // tracked by the capture_audio_. |
| StreamConfig capture_processing_format; |
| int split_rate; |
| bool echo_path_gain_change; |
| int prev_analog_mic_level; |
| float prev_pre_amp_gain; |
| int playout_volume; |
| int prev_playout_volume; |
| AudioProcessingStats stats; |
| struct KeyboardInfo { |
| void Extract(const float* const* data, const StreamConfig& stream_config); |
| size_t num_keyboard_frames = 0; |
| const float* keyboard_data = nullptr; |
| } keyboard_info; |
| } capture_ RTC_GUARDED_BY(crit_capture_); |
| |
| struct ApmCaptureNonLockedState { |
| ApmCaptureNonLockedState() |
| : capture_processing_format(kSampleRate16kHz), |
| split_rate(kSampleRate16kHz), |
| stream_delay_ms(0) {} |
| // Only the rate and samples fields of capture_processing_format_ are used |
| // because the forward processing number of channels is mutable and is |
| // tracked by the capture_audio_. |
| StreamConfig capture_processing_format; |
| int split_rate; |
| int stream_delay_ms; |
| bool echo_controller_enabled = false; |
| bool use_aec2_extended_filter = false; |
| bool use_aec2_delay_agnostic = false; |
| bool use_aec2_refined_adaptive_filter = false; |
| } capture_nonlocked_; |
| |
| struct ApmRenderState { |
| ApmRenderState(); |
| ~ApmRenderState(); |
| std::unique_ptr<AudioConverter> render_converter; |
| std::unique_ptr<AudioBuffer> render_audio; |
| } render_ RTC_GUARDED_BY(crit_render_); |
| |
| std::vector<float> aec_render_queue_buffer_ RTC_GUARDED_BY(crit_render_); |
| std::vector<float> aec_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_); |
| |
| std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(crit_render_); |
| std::vector<int16_t> aecm_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_); |
| |
| size_t agc_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_) |
| RTC_GUARDED_BY(crit_capture_) = 0; |
| std::vector<int16_t> agc_render_queue_buffer_ RTC_GUARDED_BY(crit_render_); |
| std::vector<int16_t> agc_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_); |
| |
| size_t red_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_) |
| RTC_GUARDED_BY(crit_capture_) = 0; |
| std::vector<float> red_render_queue_buffer_ RTC_GUARDED_BY(crit_render_); |
| std::vector<float> red_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_); |
| |
| RmsLevel capture_input_rms_ RTC_GUARDED_BY(crit_capture_); |
| RmsLevel capture_output_rms_ RTC_GUARDED_BY(crit_capture_); |
| int capture_rms_interval_counter_ RTC_GUARDED_BY(crit_capture_) = 0; |
| |
| // Lock protection not needed. |
| std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> |
| aec_render_signal_queue_; |
| std::unique_ptr< |
| SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> |
| aecm_render_signal_queue_; |
| std::unique_ptr< |
| SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> |
| agc_render_signal_queue_; |
| std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> |
| red_render_signal_queue_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |