Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
This commit is contained in:
Arun Raghavan
@ -1,5 +1,5 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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@ -8,81 +8,140 @@
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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_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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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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#include <list>
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#include <string>
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#include <vector>
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#include "audio_processing.h"
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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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namespace webrtc {
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namespace audioproc {
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class Event;
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} // audioproc
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class AgcManagerDirect;
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class AudioBuffer;
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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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enum {
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kSampleRate8kHz = 8000,
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kSampleRate16kHz = 16000,
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kSampleRate32kHz = 32000
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};
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explicit AudioProcessingImpl(const Config& config);
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explicit AudioProcessingImpl(int id);
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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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CriticalSectionWrapper* crit() const;
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int split_sample_rate_hz() const;
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bool was_stream_delay_set() const;
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// AudioProcessing methods.
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virtual int Initialize();
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virtual int InitializeLocked();
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virtual int set_sample_rate_hz(int rate);
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virtual int sample_rate_hz() const;
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virtual int set_num_channels(int input_channels, int output_channels);
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virtual int num_input_channels() const;
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virtual int num_output_channels() const;
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virtual int set_num_reverse_channels(int channels);
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virtual int num_reverse_channels() const;
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virtual int ProcessStream(AudioFrame* frame);
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virtual int AnalyzeReverseStream(AudioFrame* frame);
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virtual int set_stream_delay_ms(int delay);
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virtual int stream_delay_ms() const;
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virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
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virtual int StopDebugRecording();
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virtual EchoCancellation* echo_cancellation() const;
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virtual EchoControlMobile* echo_control_mobile() const;
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virtual GainControl* gain_control() const;
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virtual HighPassFilter* high_pass_filter() const;
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virtual LevelEstimator* level_estimator() const;
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virtual NoiseSuppression* noise_suppression() const;
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virtual VoiceDetection* voice_detection() const;
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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(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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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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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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// Module methods.
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virtual WebRtc_Word32 Version(WebRtc_Word8* version,
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WebRtc_UWord32& remainingBufferInBytes,
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WebRtc_UWord32& position) const;
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virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
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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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private:
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int WriteMessageToDebugFile();
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int WriteInitMessage();
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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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// TODO(ekm): Remove once all clients updated to new interface.
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int AnalyzeReverseStream(const float* const* src,
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const StreamConfig& input_config,
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const StreamConfig& output_config);
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int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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int id_;
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bool is_data_processed() const;
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bool output_copy_needed(bool is_data_processed) const;
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bool synthesis_needed(bool is_data_processed) const;
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bool analysis_needed(bool is_data_processed) const;
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bool is_rev_processed() const;
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bool rev_conversion_needed() const;
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void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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EchoCancellationImpl* echo_cancellation_;
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EchoControlMobileImpl* echo_control_mobile_;
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@ -91,27 +150,69 @@ class AudioProcessingImpl : public AudioProcessing {
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LevelEstimatorImpl* level_estimator_;
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NoiseSuppressionImpl* noise_suppression_;
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VoiceDetectionImpl* voice_detection_;
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rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
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std::list<ProcessingComponent*> component_list_;
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FileWrapper* debug_file_;
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audioproc::Event* event_msg_; // Protobuf message.
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std::string event_str_; // Memory for protobuf serialization.
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CriticalSectionWrapper* crit_;
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rtc::scoped_ptr<AudioBuffer> render_audio_;
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rtc::scoped_ptr<AudioBuffer> capture_audio_;
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rtc::scoped_ptr<AudioConverter> render_converter_;
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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// TODO(andrew): make this more graceful. Ideally we would split this stuff
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// out into a separate class with an "enabled" and "disabled" implementation.
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int WriteMessageToDebugFile();
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int WriteInitMessage();
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AudioBuffer* render_audio_;
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AudioBuffer* capture_audio_;
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// Writes Config message. If not |forced|, only writes the current config if
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// it is different from the last saved one; if |forced|, writes the config
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// regardless of the last saved.
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int WriteConfigMessage(bool forced);
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rtc::scoped_ptr<FileWrapper> debug_file_;
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rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
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std::string event_str_; // Memory for protobuf serialization.
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// Serialized string of last saved APM configuration.
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std::string last_serialized_config_;
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#endif
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// Format of processing streams at input/output call sites.
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ProcessingConfig api_format_;
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// Only the rate and samples fields of fwd_proc_format_ are used because the
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// forward processing number of channels is mutable and is tracked by the
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// capture_audio_.
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StreamConfig fwd_proc_format_;
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StreamConfig rev_proc_format_;
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int split_rate_;
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int sample_rate_hz_;
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int split_sample_rate_hz_;
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int samples_per_channel_;
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int stream_delay_ms_;
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int delay_offset_ms_;
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bool was_stream_delay_set_;
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int last_stream_delay_ms_;
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int last_aec_system_delay_ms_;
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int stream_delay_jumps_;
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int aec_system_delay_jumps_;
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int num_reverse_channels_;
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int num_input_channels_;
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int num_output_channels_;
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bool output_will_be_muted_ GUARDED_BY(crit_);
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bool key_pressed_;
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// Only set through the constructor's Config parameter.
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const bool use_new_agc_;
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rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
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int agc_startup_min_volume_;
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bool transient_suppressor_enabled_;
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rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
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const bool beamformer_enabled_;
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rtc::scoped_ptr<Beamformer<float>> beamformer_;
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const std::vector<Point> array_geometry_;
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bool intelligibility_enabled_;
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rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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