| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |
| |
| #include <list> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/modules/audio_processing/audio_buffer.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/system_wrappers/include/file_wrapper.h" |
| |
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
| // Files generated at build-time by the protobuf compiler. |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" |
| #else |
| #include "webrtc/modules/audio_processing/debug.pb.h" |
| #endif |
| #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP |
| |
| namespace webrtc { |
| |
| class AgcManagerDirect; |
| class AudioConverter; |
| |
| template<typename T> |
| class Beamformer; |
| |
| 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 Config& config); |
| // AudioProcessingImpl takes ownership of beamformer. |
| AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer); |
| virtual ~AudioProcessingImpl(); |
| 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; |
| int Initialize(const ProcessingConfig& processing_config) override; |
| void SetExtraOptions(const Config& config) override; |
| void UpdateHistogramsOnCallEnd() override; |
| int StartDebugRecording(const char filename[kMaxFilenameSize], |
| int64_t max_log_size_bytes) override; |
| int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override; |
| |
| int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override; |
| int StopDebugRecording() 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; |
| |
| // 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& reverse_input_config, |
| const StreamConfig& reverse_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 |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // 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). |
| 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_render_, crit_capture_); |
| |
| private: |
| struct ApmPublicSubmodules; |
| struct ApmPrivateSubmodules; |
| |
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP |
| // State for the debug dump. |
| struct ApmDebugDumpThreadState { |
| ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {} |
| std::unique_ptr<audioproc::Event> event_msg; // Protobuf message. |
| std::string event_str; // Memory for protobuf serialization. |
| |
| // Serialized string of last saved APM configuration. |
| std::string last_serialized_config; |
| }; |
| |
| struct ApmDebugDumpState { |
| ApmDebugDumpState() : debug_file(FileWrapper::Create()) {} |
| // Number of bytes that can still be written to the log before the maximum |
| // size is reached. A value of <= 0 indicates that no limit is used. |
| int64_t num_bytes_left_for_log_ = -1; |
| std::unique_ptr<FileWrapper> debug_file; |
| ApmDebugDumpThreadState render; |
| ApmDebugDumpThreadState capture; |
| }; |
| #endif |
| |
| // 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 MaybeInitialize(const ProcessingConfig& config) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| int MaybeInitializeRender(const ProcessingConfig& processing_config) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| int MaybeInitializeCapture(const ProcessingConfig& processing_config) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Method for checking for the need of conversion. Accesses the formats |
| // structs in a read manner but the requirement for the render lock to be held |
| // was added as it currently anyway is always called in that manner. |
| bool rev_conversion_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| bool render_check_rev_conversion_needed() const |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Methods requiring APM running in a single-threaded manner. |
| // Are called with both the render and capture locks already |
| // acquired. |
| void InitializeExperimentalAgc() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeTransient() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeBeamformer() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeIntelligibility() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeHighPassFilter() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeNoiseSuppression() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeLevelEstimator() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeVoiceDetection() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void InitializeEchoCanceller() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeGainController() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| void InitializeEchoControlMobile() |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| int InitializeLocked(const ProcessingConfig& config) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_); |
| |
| // Capture-side exclusive methods possibly running APM in a multi-threaded |
| // manner that are called with the render lock already acquired. |
| int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| bool output_copy_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| bool is_fwd_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| bool fwd_synthesis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| bool fwd_analysis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| void MaybeUpdateHistograms() 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) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| bool rev_synthesis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| bool rev_analysis_needed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_); |
| |
| // Debug dump methods that are internal and called without locks. |
| // TODO(peah): Make thread safe. |
| #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. |
| static int WriteMessageToDebugFile(FileWrapper* debug_file, |
| int64_t* filesize_limit_bytes, |
| rtc::CriticalSection* crit_debug, |
| ApmDebugDumpThreadState* debug_state); |
| int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_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) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_capture_); |
| |
| // Critical section. |
| rtc::CriticalSection crit_debug_; |
| |
| // Debug dump state. |
| ApmDebugDumpState debug_dump_; |
| #endif |
| |
| // Critical sections. |
| rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_); |
| rtc::CriticalSection crit_capture_; |
| |
| // Structs containing the pointers to the submodules. |
| std::unique_ptr<ApmPublicSubmodules> public_submodules_; |
| std::unique_ptr<ApmPrivateSubmodules> private_submodules_ |
| GUARDED_BY(crit_capture_); |
| |
| // 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}}}), |
| rev_proc_format(kSampleRate16kHz, 1) {} |
| ProcessingConfig api_format; |
| StreamConfig rev_proc_format; |
| } formats_; |
| |
| // APM constants. |
| const struct ApmConstants { |
| ApmConstants(int agc_startup_min_volume, bool use_experimental_agc) |
| : // Format of processing streams at input/output call sites. |
| agc_startup_min_volume(agc_startup_min_volume), |
| use_experimental_agc(use_experimental_agc) {} |
| int agc_startup_min_volume; |
| bool use_experimental_agc; |
| } constants_; |
| |
| struct ApmCaptureState { |
| ApmCaptureState(bool transient_suppressor_enabled, |
| const std::vector<Point>& array_geometry, |
| SphericalPointf target_direction) |
| : aec_system_delay_jumps(-1), |
| delay_offset_ms(0), |
| was_stream_delay_set(false), |
| last_stream_delay_ms(0), |
| last_aec_system_delay_ms(0), |
| stream_delay_jumps(-1), |
| output_will_be_muted(false), |
| key_pressed(false), |
| transient_suppressor_enabled(transient_suppressor_enabled), |
| array_geometry(array_geometry), |
| target_direction(target_direction), |
| fwd_proc_format(kSampleRate16kHz), |
| split_rate(kSampleRate16kHz) {} |
| int aec_system_delay_jumps; |
| int delay_offset_ms; |
| bool was_stream_delay_set; |
| int last_stream_delay_ms; |
| int last_aec_system_delay_ms; |
| int stream_delay_jumps; |
| bool output_will_be_muted; |
| bool key_pressed; |
| bool transient_suppressor_enabled; |
| std::vector<Point> array_geometry; |
| SphericalPointf target_direction; |
| std::unique_ptr<AudioBuffer> capture_audio; |
| // 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; |
| int split_rate; |
| } capture_ GUARDED_BY(crit_capture_); |
| |
| struct ApmCaptureNonLockedState { |
| ApmCaptureNonLockedState(bool beamformer_enabled, |
| bool intelligibility_enabled) |
| : fwd_proc_format(kSampleRate16kHz), |
| split_rate(kSampleRate16kHz), |
| stream_delay_ms(0), |
| beamformer_enabled(beamformer_enabled), |
| intelligibility_enabled(intelligibility_enabled) {} |
| // 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; |
| int split_rate; |
| int stream_delay_ms; |
| bool beamformer_enabled; |
| bool intelligibility_enabled; |
| } capture_nonlocked_; |
| |
| struct ApmRenderState { |
| std::unique_ptr<AudioConverter> render_converter; |
| std::unique_ptr<AudioBuffer> render_audio; |
| } render_ GUARDED_BY(crit_render_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_ |