tg2sip/webrtc_dsp/modules/audio_processing/audio_processing_impl.h

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* 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_