| Index: webrtc/modules/audio_processing/audio_processing_impl.h
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| index f8d9f7c2ba5d4bf1ee8f2be8b684728597e69048..28db7b3d05337ec04c93309445f302c5ca1eaa54 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| @@ -15,24 +15,35 @@
|
| #include <string>
|
| #include <vector>
|
|
|
| +#include "webrtc/base/criticalsection.h"
|
| #include "webrtc/base/scoped_ptr.h"
|
| #include "webrtc/base/thread_annotations.h"
|
| #include "webrtc/base/thread_checker.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/audio_processing/debug.pb.h"
|
| +#endif
|
| +#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
|
|
|
| namespace webrtc {
|
|
|
| class AgcManagerDirect;
|
| -class AudioBuffer;
|
| class AudioConverter;
|
|
|
| template<typename T>
|
| class Beamformer;
|
|
|
| -class CriticalSectionWrapper;
|
| +struct ApmPublicSubmodules;
|
| +struct ApmPrivateSubmodules;
|
| class EchoCancellationImpl;
|
| class EchoControlMobileImpl;
|
| -class FileWrapper;
|
| class GainControlImpl;
|
| class GainControlForNewAgc;
|
| class HighPassFilterImpl;
|
| @@ -49,17 +60,34 @@ namespace audioproc {
|
| class Event;
|
|
|
| } // namespace audioproc
|
| +
|
| +// State for the debug dump.
|
| +struct ApmDebugDumpThreadState {
|
| + ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
|
| + rtc::scoped_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()) {}
|
| + rtc::scoped_ptr<FileWrapper> debug_file;
|
| + ApmDebugDumpThreadState render;
|
| + ApmDebugDumpThreadState capture;
|
| +};
|
| +
|
| #endif
|
|
|
| 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();
|
| -
|
| - // AudioProcessing methods.
|
| int Initialize() override;
|
| int Initialize(int input_sample_rate_hz,
|
| int output_sample_rate_hz,
|
| @@ -69,12 +97,14 @@ class AudioProcessingImpl : public AudioProcessing {
|
| ChannelLayout reverse_layout) override;
|
| int Initialize(const ProcessingConfig& processing_config) override;
|
| void SetExtraOptions(const Config& config) override;
|
| - int proc_sample_rate_hz() const override;
|
| - int proc_split_sample_rate_hz() const override;
|
| - int num_input_channels() const override;
|
| - int num_output_channels() const override;
|
| - int num_reverse_channels() const override;
|
| - void set_output_will_be_muted(bool muted) override;
|
| + void UpdateHistogramsOnCallEnd() override;
|
| + int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
|
| + int StartDebugRecording(FILE* handle) 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,
|
| @@ -87,6 +117,14 @@ class AudioProcessingImpl : public AudioProcessing {
|
| 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 AnalyzeReverseStream(AudioFrame* frame) override;
|
| int ProcessReverseStream(AudioFrame* frame) override;
|
| int AnalyzeReverseStream(const float* const* data,
|
| @@ -97,17 +135,24 @@ class AudioProcessingImpl : public AudioProcessing {
|
| const StreamConfig& reverse_input_config,
|
| const StreamConfig& reverse_output_config,
|
| float* const* dest) override;
|
| - int set_stream_delay_ms(int delay) 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;
|
| + int num_input_channels() const override;
|
| + int num_output_channels() const override;
|
| + int num_reverse_channels() const override;
|
| int stream_delay_ms() const override;
|
| - bool was_stream_delay_set() const override;
|
| - void set_delay_offset_ms(int offset) override;
|
| - int delay_offset_ms() const override;
|
| - void set_stream_key_pressed(bool key_pressed) override;
|
| - int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
|
| - int StartDebugRecording(FILE* handle) override;
|
| - int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
|
| - int StopDebugRecording() override;
|
| - void UpdateHistogramsOnCallEnd() 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;
|
| @@ -118,113 +163,183 @@ class AudioProcessingImpl : public AudioProcessing {
|
|
|
| protected:
|
| // Overridden in a mock.
|
| - virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| + virtual int InitializeLocked()
|
| + EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
|
|
|
| private:
|
| + // 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_);
|
| + // 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_);
|
| +
|
| + // 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_);
|
| int InitializeLocked(const ProcessingConfig& config)
|
| - EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - int MaybeInitializeLocked(const ProcessingConfig& config)
|
| - EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| + 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(bool is_data_processed) const
|
| + EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
|
| + bool is_data_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
|
| + bool synthesis_needed(bool is_data_processed) const
|
| + EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
|
| + bool analysis_needed(bool is_data_processed) 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 AnalyzeReverseStream(const float* const* src,
|
| - const StreamConfig& input_config,
|
| - const StreamConfig& output_config);
|
| - int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| - int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
|
| -
|
| - bool is_data_processed() const;
|
| - bool output_copy_needed(bool is_data_processed) const;
|
| - bool synthesis_needed(bool is_data_processed) const;
|
| - bool analysis_needed(bool is_data_processed) const;
|
| - 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_);
|
| -
|
| - 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_;
|
| -
|
| - std::list<ProcessingComponent*> component_list_;
|
| - CriticalSectionWrapper* crit_;
|
| - rtc::ThreadChecker render_thread_checker_;
|
| - rtc::ThreadChecker capture_thread_checker_;
|
| - rtc::ThreadChecker signal_thread_checker_;
|
| - rtc::scoped_ptr<AudioBuffer> render_audio_;
|
| - rtc::scoped_ptr<AudioBuffer> capture_audio_;
|
| - rtc::scoped_ptr<AudioConverter> render_converter_;
|
| + 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_);
|
| + 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.
|
| - int WriteMessageToDebugFile();
|
| - int WriteInitMessage();
|
| + static int WriteMessageToDebugFile(FileWrapper* debug_file,
|
| + 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);
|
| -
|
| - rtc::scoped_ptr<FileWrapper> debug_file_;
|
| - rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
|
| - std::string event_str_; // Memory for protobuf serialization.
|
| + int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
|
| + EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
|
| +#endif
|
|
|
| - // Serialized string of last saved APM configuration.
|
| - std::string last_serialized_config_;
|
| + // Critical sections and threadcheckers.
|
| + mutable rtc::CriticalSection crit_render_ ACQUIRED_BEFORE(crit_capture_);
|
| + mutable rtc::CriticalSection crit_capture_;
|
| +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
|
| + mutable rtc::CriticalSection crit_debug_;
|
| #endif
|
| + rtc::ThreadChecker render_thread_checker_;
|
| + rtc::ThreadChecker capture_thread_checker_;
|
| + rtc::ThreadChecker signal_thread_checker_;
|
| +
|
| + // Structs containing the pointers to the submodules.
|
| + rtc::scoped_ptr<ApmPublicSubmodules> public_submodules_;
|
| + rtc::scoped_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 while holding only one of the locks.
|
| - struct SharedState {
|
| - SharedState()
|
| + // 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}}}) {}
|
| - ProcessingConfig api_format_;
|
| - } shared_state_;
|
| -
|
| - // 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_;
|
| -
|
| - 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_;
|
| -
|
| - bool output_will_be_muted_ GUARDED_BY(crit_);
|
| -
|
| - bool key_pressed_;
|
| -
|
| - // 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_;
|
| -
|
| - 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_;
|
| -
|
| - bool intelligibility_enabled_;
|
| - rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
|
| + 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,
|
| + const std::vector<Point> array_geometry,
|
| + SphericalPointf target_direction,
|
| + bool use_new_agc,
|
| + bool intelligibility_enabled,
|
| + bool beamformer_enabled)
|
| + : // Format of processing streams at input/output call sites.
|
| + agc_startup_min_volume(agc_startup_min_volume),
|
| + array_geometry(array_geometry),
|
| + target_direction(target_direction),
|
| + use_new_agc(use_new_agc),
|
| + intelligibility_enabled(intelligibility_enabled),
|
| + beamformer_enabled(beamformer_enabled) {}
|
| + int agc_startup_min_volume;
|
| + std::vector<Point> array_geometry;
|
| + SphericalPointf target_direction;
|
| + bool use_new_agc;
|
| + bool intelligibility_enabled;
|
| + bool beamformer_enabled;
|
| + } constants_;
|
| +
|
| + struct ApmCaptureState {
|
| + ApmCaptureState(bool transient_suppressor_enabled)
|
| + : 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),
|
| + 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;
|
| + rtc::scoped_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()
|
| + : fwd_proc_format(kSampleRate16kHz),
|
| + split_rate(kSampleRate16kHz),
|
| + stream_delay_ms(0) {}
|
| + // 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;
|
| + } capture_nonlocked_;
|
| +
|
| + struct ApmRenderState {
|
| + rtc::scoped_ptr<AudioConverter> render_converter;
|
| + rtc::scoped_ptr<AudioBuffer> render_audio;
|
| + } render_ GUARDED_BY(crit_render_);
|
| +
|
| +// Debug dump state.
|
| +#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
|
| + ApmDebugDumpState debug_dump_;
|
| +#endif
|
| };
|
|
|
| } // namespace webrtc
|
|
|