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/*
* 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;
class NonlinearBeamformer;
class AudioProcessingImpl : public AudioProcessing {
public:
// Methods forcing APM to run in a single-threaded manner.
// Acquires both the render and capture locks.
explicit AudioProcessingImpl(const webrtc::Config& config);
// AudioProcessingImpl takes ownership of beamformer.
AudioProcessingImpl(const webrtc::Config& config,
NonlinearBeamformer* beamformer);
~AudioProcessingImpl() override;
int Initialize() override;
int Initialize(int capture_input_sample_rate_hz,
int capture_output_sample_rate_hz,
int render_sample_rate_hz,
ChannelLayout capture_input_layout,
ChannelLayout capture_output_layout,
ChannelLayout render_input_layout) override;
int Initialize(const ProcessingConfig& processing_config) override;
void ApplyConfig(const AudioProcessing::Config& config) override;
void SetExtraOptions(const webrtc::Config& config) override;
void UpdateHistogramsOnCallEnd() override;
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& input_config,
const StreamConfig& output_config,
float* const* dest) override;
// Methods only accessed from APM submodules or
// from AudioProcessing tests in a single-threaded manner.
// Hence there is no need for locks in these.
int proc_sample_rate_hz() const override;
int proc_split_sample_rate_hz() const override;
size_t num_input_channels() const override;
size_t num_proc_channels() const override;
size_t num_output_channels() const override;
size_t num_reverse_channels() const override;
int stream_delay_ms() const override;
bool was_stream_delay_set() const override
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;
class ApmSubmoduleStates {
public:
ApmSubmoduleStates();
// Updates the submodule state and returns true if it has changed.
bool Update(bool high_pass_filter_enabled,
bool echo_canceller_enabled,
bool mobile_echo_controller_enabled,
bool noise_suppressor_enabled,
bool intelligibility_enhancer_enabled,
bool beamformer_enabled,
bool adaptive_gain_controller_enabled,
bool level_controller_enabled,
bool voice_activity_detector_enabled,
bool level_estimator_enabled,
bool transient_suppressor_enabled);
bool CaptureMultiBandSubModulesActive() const;
bool CaptureMultiBandProcessingActive() const;
bool RenderMultiBandSubModulesActive() const;
bool RenderMultiBandProcessingActive() const;
private:
bool high_pass_filter_enabled_ = false;
bool echo_canceller_enabled_ = false;
bool mobile_echo_controller_enabled_ = false;
bool noise_suppressor_enabled_ = false;
bool intelligibility_enhancer_enabled_ = false;
bool beamformer_enabled_ = false;
bool adaptive_gain_controller_enabled_ = false;
bool level_controller_enabled_ = false;
bool level_estimator_enabled_ = false;
bool voice_activity_detector_enabled_ = false;
bool transient_suppressor_enabled_ = false;
bool first_update_ = true;
};
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// State for the debug dump.
struct ApmDebugDumpThreadState {
ApmDebugDumpThreadState();
~ApmDebugDumpThreadState();
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();
~ApmDebugDumpState();
// 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, bool force_initialization)
EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
int MaybeInitializeRender(const ProcessingConfig& processing_config)
EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
int MaybeInitializeCapture(const ProcessingConfig& processing_config,
bool force_initialization)
EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
// Method for updating the state keeping track of the active submodules.
// Returns a bool indicating whether the state has changed.
bool UpdateActiveSubmoduleStates() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// Methods requiring APM running in a single-threaded manner.
// Are called with both the render and capture locks already
// acquired.
void InitializeTransient()
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_render_, crit_capture_);
void InitializeLevelController() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// Capture-side exclusive methods possibly running APM in a multi-threaded
// manner that are called with the render lock already acquired.
int ProcessCaptureStreamLocked() 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_);
int ProcessRenderStreamLocked() 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_;
// Class containing information about what submodules are active.
ApmSubmoduleStates submodule_states_;
// Structs containing the pointers to the submodules.
std::unique_ptr<ApmPublicSubmodules> public_submodules_;
std::unique_ptr<ApmPrivateSubmodules> private_submodules_
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}}}),
render_processing_format(kSampleRate16kHz, 1) {}
ProcessingConfig api_format;
StreamConfig render_processing_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);
~ApmCaptureState();
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 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;
} capture_ GUARDED_BY(crit_capture_);
struct ApmCaptureNonLockedState {
ApmCaptureNonLockedState(bool beamformer_enabled,
bool intelligibility_enabled)
: capture_processing_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz),
stream_delay_ms(0),
beamformer_enabled(beamformer_enabled),
intelligibility_enabled(intelligibility_enabled) {}
// 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 beamformer_enabled;
bool intelligibility_enabled;
bool level_controller_enabled = false;
} capture_nonlocked_;
struct ApmRenderState {
ApmRenderState();
~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_