1
0
mirror of https://github.com/danog/libtgvoip.git synced 2024-12-02 17:51:06 +01:00
libtgvoip/webrtc_dsp/modules/audio_processing/audio_processing_impl.h

456 lines
20 KiB
C
Raw Normal View History

/*
* 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 MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#include <list>
#include <memory>
#include <vector>
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/render_queue_item_verifier.h"
#include "modules/audio_processing/rms_level.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/function_view.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/swap_queue.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class ApmDataDumper;
class AudioConverter;
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 capture post processor.
AudioProcessingImpl(const webrtc::Config& config,
std::unique_ptr<CustomProcessing> capture_post_processor,
std::unique_ptr<CustomProcessing> render_pre_processor,
std::unique_ptr<EchoControlFactory> echo_control_factory,
rtc::scoped_refptr<EchoDetector> echo_detector,
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer);
~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;
void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
void DetachAecDump() override;
void AttachPlayoutAudioGenerator(
std::unique_ptr<AudioGenerator> audio_generator) override;
void DetachPlayoutAudioGenerator() override;
void SetRuntimeSetting(RuntimeSetting setting) 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
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
AudioProcessingStats GetStatistics(bool has_remote_tracks) const override;
// 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).
GainControl* gain_control() const override;
LevelEstimator* level_estimator() const override;
NoiseSuppression* noise_suppression() const override;
VoiceDetection* voice_detection() const override;
// TODO(peah): Remove MutateConfig once the new API allows that.
void MutateConfig(rtc::FunctionView<void(AudioProcessing::Config*)> mutator);
AudioProcessing::Config GetConfig() const override;
protected:
// Overridden in a mock.
virtual int InitializeLocked()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
private:
// TODO(peah): These friend classes should be removed as soon as the new
// parameter setting scheme allows.
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
// Class providing thread-safe message pipe functionality for
// |runtime_settings_|.
class RuntimeSettingEnqueuer {
public:
explicit RuntimeSettingEnqueuer(
SwapQueue<RuntimeSetting>* runtime_settings);
~RuntimeSettingEnqueuer();
void Enqueue(RuntimeSetting setting);
private:
SwapQueue<RuntimeSetting>& runtime_settings_;
};
struct ApmPublicSubmodules;
struct ApmPrivateSubmodules;
std::unique_ptr<ApmDataDumper> data_dumper_;
static int instance_count_;
SwapQueue<RuntimeSetting> capture_runtime_settings_;
SwapQueue<RuntimeSetting> render_runtime_settings_;
RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
// EchoControl factory.
std::unique_ptr<EchoControlFactory> echo_control_factory_;
class ApmSubmoduleStates {
public:
ApmSubmoduleStates(bool capture_post_processor_enabled,
bool render_pre_processor_enabled,
bool capture_analyzer_enabled);
// 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 residual_echo_detector_enabled,
bool noise_suppressor_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
bool pre_amplifier_enabled,
bool echo_controller_enabled,
bool voice_activity_detector_enabled,
bool level_estimator_enabled,
bool transient_suppressor_enabled);
bool CaptureMultiBandSubModulesActive() const;
bool CaptureMultiBandProcessingActive() const;
bool CaptureFullBandProcessingActive() const;
bool CaptureAnalyzerActive() const;
bool RenderMultiBandSubModulesActive() const;
bool RenderFullBandProcessingActive() const;
bool RenderMultiBandProcessingActive() const;
bool LowCutFilteringRequired() const;
private:
const bool capture_post_processor_enabled_ = false;
const bool render_pre_processor_enabled_ = false;
const bool capture_analyzer_enabled_ = false;
bool high_pass_filter_enabled_ = false;
bool echo_canceller_enabled_ = false;
bool mobile_echo_controller_enabled_ = false;
bool residual_echo_detector_enabled_ = false;
bool noise_suppressor_enabled_ = false;
bool adaptive_gain_controller_enabled_ = false;
bool gain_controller2_enabled_ = false;
bool pre_amplifier_enabled_ = false;
bool echo_controller_enabled_ = false;
bool level_estimator_enabled_ = false;
bool voice_activity_detector_enabled_ = false;
bool transient_suppressor_enabled_ = false;
bool first_update_ = true;
};
// 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)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
int MaybeInitializeRender(const ProcessingConfig& processing_config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
int MaybeInitializeCapture(const ProcessingConfig& processing_config,
bool force_initialization)
RTC_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()
RTC_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()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
int InitializeLocked(const ProcessingConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
void InitializeResidualEchoDetector()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
void InitializeLowCutFilter() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeEchoController() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializePreAmplifier() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeAnalyzer() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
// Empties and handles the respective RuntimeSetting queues.
void HandleCaptureRuntimeSettings()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void HandleRenderRuntimeSettings() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
void EmptyQueuedRenderAudio();
void AllocateRenderQueue()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
void QueueBandedRenderAudio(AudioBuffer* audio)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
void QueueNonbandedRenderAudio(AudioBuffer* audio)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
// 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(crit_capture_);
void MaybeUpdateHistograms() RTC_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)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
int ProcessRenderStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
// 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(crit_capture_);
// Notifies attached AecDump of current configuration and capture data.
void RecordUnprocessedCaptureStream(const float* const* capture_stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void RecordUnprocessedCaptureStream(const AudioFrame& capture_frame)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// 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(crit_capture_);
void RecordProcessedCaptureStream(const AudioFrame& processed_capture_frame)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// Notifies attached AecDump about current state (delay, drift, etc).
void RecordAudioProcessingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
// 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_;
// Hold the last config written with AecDump for avoiding writing
// the same config twice.
InternalAPMConfig apm_config_for_aec_dump_ RTC_GUARDED_BY(crit_capture_);
// Critical sections.
rtc::CriticalSection crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_);
rtc::CriticalSection crit_capture_;
// Struct containing the Config specifying the behavior of APM.
AudioProcessing::Config config_;
// 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_;
// 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,
int agc_clipped_level_min,
bool use_experimental_agc,
bool use_experimental_agc_agc2_level_estimation,
bool use_experimental_agc_agc2_digital_adaptive,
bool use_experimental_agc_process_before_aec)
: // Format of processing streams at input/output call sites.
agc_startup_min_volume(agc_startup_min_volume),
agc_clipped_level_min(agc_clipped_level_min),
use_experimental_agc(use_experimental_agc),
use_experimental_agc_agc2_level_estimation(
use_experimental_agc_agc2_level_estimation),
use_experimental_agc_agc2_digital_adaptive(
use_experimental_agc_agc2_digital_adaptive),
use_experimental_agc_process_before_aec(
use_experimental_agc_process_before_aec) {}
int agc_startup_min_volume;
int agc_clipped_level_min;
bool use_experimental_agc;
bool use_experimental_agc_agc2_level_estimation;
bool use_experimental_agc_agc2_digital_adaptive;
bool use_experimental_agc_process_before_aec;
} constants_;
struct ApmCaptureState {
ApmCaptureState(bool transient_suppressor_enabled);
~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::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;
bool echo_path_gain_change;
int prev_analog_mic_level;
float prev_pre_amp_gain;
} capture_ RTC_GUARDED_BY(crit_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(crit_render_);
size_t aec_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_)
RTC_GUARDED_BY(crit_capture_) = 0;
std::vector<float> aec_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<float> aec_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);
size_t aecm_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_)
RTC_GUARDED_BY(crit_capture_) = 0;
std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<int16_t> aecm_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);
size_t agc_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_)
RTC_GUARDED_BY(crit_capture_) = 0;
std::vector<int16_t> agc_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<int16_t> agc_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);
size_t red_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_)
RTC_GUARDED_BY(crit_capture_) = 0;
std::vector<float> red_render_queue_buffer_ RTC_GUARDED_BY(crit_render_);
std::vector<float> red_capture_queue_buffer_ RTC_GUARDED_BY(crit_capture_);
RmsLevel capture_input_rms_ RTC_GUARDED_BY(crit_capture_);
RmsLevel capture_output_rms_ RTC_GUARDED_BY(crit_capture_);
int capture_rms_interval_counter_ RTC_GUARDED_BY(crit_capture_) = 0;
// Lock protection not needed.
std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
aec_render_signal_queue_;
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 // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_