Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
@ -8,212 +8,514 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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#include <stdio.h>
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#include <list>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "api/function_view.h"
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#include "modules/audio_processing/aec3/echo_canceller3.h"
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#include "modules/audio_processing/agc/agc_manager_direct.h"
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#include "modules/audio_processing/agc/gain_control.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/echo_control_mobile_impl.h"
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#include "modules/audio_processing/gain_control_impl.h"
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#include "modules/audio_processing/gain_controller2.h"
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#include "modules/audio_processing/high_pass_filter.h"
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#include "modules/audio_processing/include/aec_dump.h"
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#include "modules/audio_processing/include/audio_frame_proxies.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/include/audio_processing_statistics.h"
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#include "modules/audio_processing/level_estimator.h"
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#include "modules/audio_processing/ns/noise_suppressor.h"
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#include "modules/audio_processing/optionally_built_submodule_creators.h"
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#include "modules/audio_processing/render_queue_item_verifier.h"
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#include "modules/audio_processing/residual_echo_detector.h"
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#include "modules/audio_processing/rms_level.h"
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#include "modules/audio_processing/transient/transient_suppressor.h"
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#include "modules/audio_processing/voice_detection.h"
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#include "rtc_base/gtest_prod_util.h"
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#include "rtc_base/ignore_wundef.h"
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#include "rtc_base/swap_queue.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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class AgcManagerDirect;
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class AudioBuffer;
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class ApmDataDumper;
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class AudioConverter;
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template<typename T>
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class Beamformer;
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class CriticalSectionWrapper;
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class EchoCancellationImpl;
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class EchoControlMobileImpl;
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class FileWrapper;
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class GainControlImpl;
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class GainControlForNewAgc;
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class HighPassFilterImpl;
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class LevelEstimatorImpl;
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class NoiseSuppressionImpl;
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class ProcessingComponent;
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class TransientSuppressor;
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class VoiceDetectionImpl;
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class IntelligibilityEnhancer;
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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namespace audioproc {
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class Event;
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} // namespace audioproc
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#endif
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class AudioProcessingImpl : public AudioProcessing {
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public:
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explicit AudioProcessingImpl(const Config& config);
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// AudioProcessingImpl takes ownership of beamformer.
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AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
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virtual ~AudioProcessingImpl();
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// AudioProcessing methods.
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// Methods forcing APM to run in a single-threaded manner.
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// Acquires both the render and capture locks.
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explicit AudioProcessingImpl(const webrtc::Config& config);
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// AudioProcessingImpl takes ownership of capture post processor.
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AudioProcessingImpl(const webrtc::Config& config,
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std::unique_ptr<CustomProcessing> capture_post_processor,
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std::unique_ptr<CustomProcessing> render_pre_processor,
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std::unique_ptr<EchoControlFactory> echo_control_factory,
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rtc::scoped_refptr<EchoDetector> echo_detector,
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std::unique_ptr<CustomAudioAnalyzer> capture_analyzer);
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~AudioProcessingImpl() override;
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int Initialize() override;
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int Initialize(int input_sample_rate_hz,
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int output_sample_rate_hz,
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int reverse_sample_rate_hz,
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ChannelLayout input_layout,
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ChannelLayout output_layout,
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ChannelLayout reverse_layout) override;
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int Initialize(int capture_input_sample_rate_hz,
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int capture_output_sample_rate_hz,
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int render_sample_rate_hz,
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ChannelLayout capture_input_layout,
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ChannelLayout capture_output_layout,
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ChannelLayout render_input_layout) override;
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int Initialize(const ProcessingConfig& processing_config) override;
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void SetExtraOptions(const Config& config) override;
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int proc_sample_rate_hz() const override;
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int proc_split_sample_rate_hz() const override;
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int num_input_channels() const override;
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int num_output_channels() const override;
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int num_reverse_channels() const override;
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void set_output_will_be_muted(bool muted) override;
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int ProcessStream(AudioFrame* frame) override;
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int ProcessStream(const float* const* src,
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size_t samples_per_channel,
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int input_sample_rate_hz,
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ChannelLayout input_layout,
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int output_sample_rate_hz,
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ChannelLayout output_layout,
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float* const* dest) override;
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void ApplyConfig(const AudioProcessing::Config& config) override;
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bool CreateAndAttachAecDump(const std::string& file_name,
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int64_t max_log_size_bytes,
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rtc::TaskQueue* worker_queue) override;
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bool CreateAndAttachAecDump(FILE* handle,
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int64_t max_log_size_bytes,
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rtc::TaskQueue* worker_queue) override;
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// TODO(webrtc:5298) Deprecated variant.
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void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
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void DetachAecDump() override;
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void SetRuntimeSetting(RuntimeSetting setting) override;
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// Capture-side exclusive methods possibly running APM in a
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// multi-threaded manner. Acquire the capture lock.
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int ProcessStream(const int16_t* const src,
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const StreamConfig& input_config,
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const StreamConfig& output_config,
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int16_t* const dest) override;
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int ProcessStream(const float* const* src,
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const StreamConfig& input_config,
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const StreamConfig& output_config,
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float* const* dest) override;
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int AnalyzeReverseStream(AudioFrame* frame) override;
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int ProcessReverseStream(AudioFrame* frame) override;
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int AnalyzeReverseStream(const float* const* data,
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size_t samples_per_channel,
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int sample_rate_hz,
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ChannelLayout layout) override;
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int ProcessReverseStream(const float* const* src,
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const StreamConfig& reverse_input_config,
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const StreamConfig& reverse_output_config,
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float* const* dest) override;
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bool GetLinearAecOutput(
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rtc::ArrayView<std::array<float, 160>> linear_output) const override;
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void set_output_will_be_muted(bool muted) override;
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int set_stream_delay_ms(int delay) override;
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int stream_delay_ms() const override;
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bool was_stream_delay_set() const override;
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void set_delay_offset_ms(int offset) override;
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int delay_offset_ms() const override;
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void set_stream_key_pressed(bool key_pressed) override;
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int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
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int StartDebugRecording(FILE* handle) override;
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int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
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int StopDebugRecording() override;
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void UpdateHistogramsOnCallEnd() override;
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EchoCancellation* echo_cancellation() const override;
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EchoControlMobile* echo_control_mobile() const override;
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GainControl* gain_control() const override;
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HighPassFilter* high_pass_filter() const override;
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LevelEstimator* level_estimator() const override;
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NoiseSuppression* noise_suppression() const override;
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VoiceDetection* voice_detection() const override;
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void set_stream_analog_level(int level) override;
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int recommended_stream_analog_level() const
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RTC_LOCKS_EXCLUDED(mutex_capture_) override;
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// Render-side exclusive methods possibly running APM in a
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// multi-threaded manner. Acquire the render lock.
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int ProcessReverseStream(const int16_t* const src,
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const StreamConfig& input_config,
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const StreamConfig& output_config,
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int16_t* const dest) override;
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int AnalyzeReverseStream(const float* const* data,
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const StreamConfig& reverse_config) override;
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int ProcessReverseStream(const float* const* src,
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const StreamConfig& input_config,
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const StreamConfig& output_config,
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float* const* dest) override;
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// Methods only accessed from APM submodules or
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// from AudioProcessing tests in a single-threaded manner.
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// Hence there is no need for locks in these.
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int proc_sample_rate_hz() const override;
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int proc_split_sample_rate_hz() const override;
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size_t num_input_channels() const override;
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size_t num_proc_channels() const override;
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size_t num_output_channels() const override;
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size_t num_reverse_channels() const override;
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int stream_delay_ms() const override;
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AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
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return GetStatistics();
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}
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AudioProcessingStats GetStatistics() override {
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return stats_reporter_.GetStatistics();
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}
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// TODO(peah): Remove MutateConfig once the new API allows that.
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void MutateConfig(rtc::FunctionView<void(AudioProcessing::Config*)> mutator);
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AudioProcessing::Config GetConfig() const override;
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protected:
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// Overridden in a mock.
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virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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virtual void InitializeLocked()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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private:
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// TODO(peah): These friend classes should be removed as soon as the new
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// parameter setting scheme allows.
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FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
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FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
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FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
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FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
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ToggleTransientSuppressor);
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FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
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ReinitializeTransientSuppressor);
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FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
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BitexactWithDisabledModules);
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int recommended_stream_analog_level_locked() const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void OverrideSubmoduleCreationForTesting(
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const ApmSubmoduleCreationOverrides& overrides);
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// Class providing thread-safe message pipe functionality for
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// |runtime_settings_|.
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class RuntimeSettingEnqueuer {
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public:
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explicit RuntimeSettingEnqueuer(
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SwapQueue<RuntimeSetting>* runtime_settings);
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~RuntimeSettingEnqueuer();
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void Enqueue(RuntimeSetting setting);
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private:
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SwapQueue<RuntimeSetting>& runtime_settings_;
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};
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std::unique_ptr<ApmDataDumper> data_dumper_;
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static int instance_count_;
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const bool use_setup_specific_default_aec3_config_;
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SwapQueue<RuntimeSetting> capture_runtime_settings_;
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SwapQueue<RuntimeSetting> render_runtime_settings_;
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RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
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RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
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// EchoControl factory.
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std::unique_ptr<EchoControlFactory> echo_control_factory_;
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class SubmoduleStates {
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public:
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SubmoduleStates(bool capture_post_processor_enabled,
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bool render_pre_processor_enabled,
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bool capture_analyzer_enabled);
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// Updates the submodule state and returns true if it has changed.
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bool Update(bool high_pass_filter_enabled,
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bool mobile_echo_controller_enabled,
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bool residual_echo_detector_enabled,
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bool noise_suppressor_enabled,
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bool adaptive_gain_controller_enabled,
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bool gain_controller2_enabled,
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bool pre_amplifier_enabled,
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bool echo_controller_enabled,
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bool voice_detector_enabled,
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bool transient_suppressor_enabled);
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bool CaptureMultiBandSubModulesActive() const;
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bool CaptureMultiBandProcessingPresent() const;
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bool CaptureMultiBandProcessingActive(bool ec_processing_active) const;
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bool CaptureFullBandProcessingActive() const;
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bool CaptureAnalyzerActive() const;
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bool RenderMultiBandSubModulesActive() const;
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bool RenderFullBandProcessingActive() const;
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bool RenderMultiBandProcessingActive() const;
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bool HighPassFilteringRequired() const;
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private:
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const bool capture_post_processor_enabled_ = false;
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const bool render_pre_processor_enabled_ = false;
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const bool capture_analyzer_enabled_ = false;
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bool high_pass_filter_enabled_ = false;
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bool mobile_echo_controller_enabled_ = false;
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bool residual_echo_detector_enabled_ = false;
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bool noise_suppressor_enabled_ = false;
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bool adaptive_gain_controller_enabled_ = false;
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bool gain_controller2_enabled_ = false;
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bool pre_amplifier_enabled_ = false;
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bool echo_controller_enabled_ = false;
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bool voice_detector_enabled_ = false;
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bool transient_suppressor_enabled_ = false;
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bool first_update_ = true;
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};
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// Methods for modifying the formats struct that is used by both
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// the render and capture threads. The check for whether modifications are
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// needed is done while holding a single lock only, thereby avoiding that the
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// capture thread blocks the render thread.
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// Called by render: Holds the render lock when reading the format struct and
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// acquires both locks if reinitialization is required.
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int MaybeInitializeRender(const ProcessingConfig& processing_config)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
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// Called by capture: Holds the capture lock when reading the format struct
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// and acquires both locks if reinitialization is needed.
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int MaybeInitializeCapture(const StreamConfig& input_config,
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const StreamConfig& output_config);
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// Method for updating the state keeping track of the active submodules.
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// Returns a bool indicating whether the state has changed.
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bool UpdateActiveSubmoduleStates()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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// Methods requiring APM running in a single-threaded manner, requiring both
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// the render and capture lock to be acquired.
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int InitializeLocked(const ProcessingConfig& config)
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EXCLUSIVE_LOCKS_REQUIRED(crit_);
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int MaybeInitializeLocked(const ProcessingConfig& config)
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EXCLUSIVE_LOCKS_REQUIRED(crit_);
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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void InitializeResidualEchoDetector()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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void InitializeEchoController()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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// Initializations of capture-only submodules, requiring the capture lock
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// already acquired.
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void InitializeHighPassFilter(bool forced_reset)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeVoiceDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeTransientSuppressor()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeNoiseSuppressor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializePreAmplifier() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeAnalyzer() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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// Initializations of render-only submodules, requiring the render lock
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// already acquired.
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void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
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// Sample rate used for the fullband processing.
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int proc_fullband_sample_rate_hz() const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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||||
|
||||
// Empties and handles the respective RuntimeSetting queues.
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void HandleCaptureRuntimeSettings()
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void HandleRenderRuntimeSettings()
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||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
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||||
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||||
void EmptyQueuedRenderAudio() RTC_LOCKS_EXCLUDED(mutex_capture_);
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||||
void EmptyQueuedRenderAudioLocked()
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
|
||||
void AllocateRenderQueue()
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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||||
void QueueBandedRenderAudio(AudioBuffer* audio)
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
|
||||
void QueueNonbandedRenderAudio(AudioBuffer* audio)
|
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
|
||||
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||||
// 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(mutex_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 AnalyzeReverseStream(const float* const* src,
|
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const StreamConfig& input_config,
|
||||
const StreamConfig& output_config);
|
||||
int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
int AnalyzeReverseStreamLocked(const float* const* src,
|
||||
const StreamConfig& input_config,
|
||||
const StreamConfig& output_config)
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
|
||||
int ProcessRenderStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
|
||||
|
||||
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;
|
||||
bool is_rev_processed() const;
|
||||
bool rev_conversion_needed() const;
|
||||
void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
||||
// 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(mutex_capture_);
|
||||
|
||||
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_;
|
||||
// Notifies attached AecDump of current configuration and capture data.
|
||||
void RecordUnprocessedCaptureStream(const float* const* capture_stream)
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
|
||||
|
||||
std::list<ProcessingComponent*> component_list_;
|
||||
CriticalSectionWrapper* crit_;
|
||||
rtc::scoped_ptr<AudioBuffer> render_audio_;
|
||||
rtc::scoped_ptr<AudioBuffer> capture_audio_;
|
||||
rtc::scoped_ptr<AudioConverter> render_converter_;
|
||||
#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();
|
||||
void RecordUnprocessedCaptureStream(const int16_t* const data,
|
||||
const StreamConfig& config)
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
|
||||
|
||||
// Writes Config message. 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.
|
||||
int WriteConfigMessage(bool forced);
|
||||
// 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(mutex_capture_);
|
||||
|
||||
rtc::scoped_ptr<FileWrapper> debug_file_;
|
||||
rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
|
||||
std::string event_str_; // Memory for protobuf serialization.
|
||||
void RecordProcessedCaptureStream(const int16_t* const data,
|
||||
const StreamConfig& config)
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
|
||||
|
||||
// Serialized string of last saved APM configuration.
|
||||
std::string last_serialized_config_;
|
||||
#endif
|
||||
// Notifies attached AecDump about current state (delay, drift, etc).
|
||||
void RecordAudioProcessingState()
|
||||
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
|
||||
|
||||
// Format of processing streams at input/output call sites.
|
||||
ProcessingConfig api_format_;
|
||||
// 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_;
|
||||
|
||||
// Only the rate and samples fields of fwd_proc_format_ are used because the
|
||||
// forward processing number of channels is mutable and is tracked by the
|
||||
// capture_audio_.
|
||||
StreamConfig fwd_proc_format_;
|
||||
StreamConfig rev_proc_format_;
|
||||
int split_rate_;
|
||||
// Hold the last config written with AecDump for avoiding writing
|
||||
// the same config twice.
|
||||
InternalAPMConfig apm_config_for_aec_dump_ RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
int stream_delay_ms_;
|
||||
int delay_offset_ms_;
|
||||
bool was_stream_delay_set_;
|
||||
int last_stream_delay_ms_;
|
||||
int last_aec_system_delay_ms_;
|
||||
int stream_delay_jumps_;
|
||||
int aec_system_delay_jumps_;
|
||||
// Critical sections.
|
||||
mutable Mutex mutex_render_ RTC_ACQUIRED_BEFORE(mutex_capture_);
|
||||
mutable Mutex mutex_capture_;
|
||||
|
||||
bool output_will_be_muted_ GUARDED_BY(crit_);
|
||||
// Struct containing the Config specifying the behavior of APM.
|
||||
AudioProcessing::Config config_;
|
||||
|
||||
bool key_pressed_;
|
||||
// Overrides for testing the exclusion of some submodules from the build.
|
||||
ApmSubmoduleCreationOverrides submodule_creation_overrides_
|
||||
RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
// 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_;
|
||||
// Class containing information about what submodules are active.
|
||||
SubmoduleStates submodule_states_;
|
||||
|
||||
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 SphericalPointf target_direction_;
|
||||
// Struct containing the pointers to the submodules.
|
||||
struct Submodules {
|
||||
Submodules(std::unique_ptr<CustomProcessing> capture_post_processor,
|
||||
std::unique_ptr<CustomProcessing> render_pre_processor,
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector,
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer)
|
||||
: echo_detector(std::move(echo_detector)),
|
||||
capture_post_processor(std::move(capture_post_processor)),
|
||||
render_pre_processor(std::move(render_pre_processor)),
|
||||
capture_analyzer(std::move(capture_analyzer)) {}
|
||||
// Accessed internally from capture or during initialization.
|
||||
std::unique_ptr<AgcManagerDirect> agc_manager;
|
||||
std::unique_ptr<GainControlImpl> gain_control;
|
||||
std::unique_ptr<GainController2> gain_controller2;
|
||||
std::unique_ptr<HighPassFilter> high_pass_filter;
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector;
|
||||
std::unique_ptr<EchoControl> echo_controller;
|
||||
std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
|
||||
std::unique_ptr<NoiseSuppressor> noise_suppressor;
|
||||
std::unique_ptr<TransientSuppressor> transient_suppressor;
|
||||
std::unique_ptr<CustomProcessing> capture_post_processor;
|
||||
std::unique_ptr<CustomProcessing> render_pre_processor;
|
||||
std::unique_ptr<GainApplier> pre_amplifier;
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
|
||||
std::unique_ptr<LevelEstimator> output_level_estimator;
|
||||
std::unique_ptr<VoiceDetection> voice_detector;
|
||||
} submodules_;
|
||||
|
||||
bool intelligibility_enabled_;
|
||||
rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
|
||||
// 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(bool multi_channel_render_support,
|
||||
bool multi_channel_capture_support,
|
||||
bool enforce_split_band_hpf)
|
||||
: multi_channel_render_support(multi_channel_render_support),
|
||||
multi_channel_capture_support(multi_channel_capture_support),
|
||||
enforce_split_band_hpf(enforce_split_band_hpf) {}
|
||||
bool multi_channel_render_support;
|
||||
bool multi_channel_capture_support;
|
||||
bool enforce_split_band_hpf;
|
||||
} constants_;
|
||||
|
||||
struct ApmCaptureState {
|
||||
ApmCaptureState();
|
||||
~ApmCaptureState();
|
||||
bool was_stream_delay_set;
|
||||
bool output_will_be_muted;
|
||||
bool key_pressed;
|
||||
std::unique_ptr<AudioBuffer> capture_audio;
|
||||
std::unique_ptr<AudioBuffer> capture_fullband_audio;
|
||||
std::unique_ptr<AudioBuffer> linear_aec_output;
|
||||
// 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;
|
||||
int cached_stream_analog_level_ = 0;
|
||||
} capture_ RTC_GUARDED_BY(mutex_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;
|
||||
} capture_nonlocked_;
|
||||
|
||||
struct ApmRenderState {
|
||||
ApmRenderState();
|
||||
~ApmRenderState();
|
||||
std::unique_ptr<AudioConverter> render_converter;
|
||||
std::unique_ptr<AudioBuffer> render_audio;
|
||||
} render_ RTC_GUARDED_BY(mutex_render_);
|
||||
|
||||
// Class for statistics reporting. The class is thread-safe and no lock is
|
||||
// needed when accessing it.
|
||||
class ApmStatsReporter {
|
||||
public:
|
||||
ApmStatsReporter();
|
||||
~ApmStatsReporter();
|
||||
|
||||
// Returns the most recently reported statistics.
|
||||
AudioProcessingStats GetStatistics();
|
||||
|
||||
// Update the cached statistics.
|
||||
void UpdateStatistics(const AudioProcessingStats& new_stats);
|
||||
|
||||
private:
|
||||
Mutex mutex_stats_;
|
||||
AudioProcessingStats cached_stats_ RTC_GUARDED_BY(mutex_stats_);
|
||||
SwapQueue<AudioProcessingStats> stats_message_queue_;
|
||||
} stats_reporter_;
|
||||
|
||||
std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
|
||||
std::vector<int16_t> aecm_capture_queue_buffer_
|
||||
RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
size_t agc_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
|
||||
RTC_GUARDED_BY(mutex_capture_) = 0;
|
||||
std::vector<int16_t> agc_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
|
||||
std::vector<int16_t> agc_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
size_t red_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
|
||||
RTC_GUARDED_BY(mutex_capture_) = 0;
|
||||
std::vector<float> red_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
|
||||
std::vector<float> red_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
RmsLevel capture_input_rms_ RTC_GUARDED_BY(mutex_capture_);
|
||||
RmsLevel capture_output_rms_ RTC_GUARDED_BY(mutex_capture_);
|
||||
int capture_rms_interval_counter_ RTC_GUARDED_BY(mutex_capture_) = 0;
|
||||
|
||||
// Lock protection not needed.
|
||||
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 // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
|
||||
#endif // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
|
||||
|
Reference in New Issue
Block a user