AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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164
webrtc/modules/audio_processing/agc/agc_manager_direct.h
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@ -8,29 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#ifndef MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#define MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
#include <memory>
#include "absl/types/optional.h"
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/gtest_prod_util.h"
namespace webrtc {
class AudioFrame;
class DebugFile;
class MonoAgc;
class GainControl;
// Callbacks that need to be injected into AgcManagerDirect to read and control
// the volume values. This is done to remove the VoiceEngine dependency in
// AgcManagerDirect.
// TODO(aluebs): Remove VolumeCallbacks.
class VolumeCallbacks {
public:
virtual ~VolumeCallbacks() {}
virtual void SetMicVolume(int volume) = 0;
virtual int GetMicVolume() = 0;
};
// Direct interface to use AGC to set volume and compression values.
// AudioProcessing uses this interface directly to integrate the callback-less
// AGC.
@ -42,30 +35,105 @@ class AgcManagerDirect final {
// responsible for processing the audio using it after the call to Process.
// The operating range of startup_min_level is [12, 255] and any input value
// outside that range will be clamped.
AgcManagerDirect(GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level);
// Dependency injection for testing. Don't delete |agc| as the memory is owned
// by the manager.
AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level);
~AgcManagerDirect();
AgcManagerDirect(int num_capture_channels,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int sample_rate_hz);
int Initialize();
void AnalyzePreProcess(int16_t* audio,
int num_channels,
size_t samples_per_channel);
void Process(const int16_t* audio, size_t length, int sample_rate_hz);
~AgcManagerDirect();
AgcManagerDirect(const AgcManagerDirect&) = delete;
AgcManagerDirect& operator=(const AgcManagerDirect&) = delete;
void Initialize();
void SetupDigitalGainControl(GainControl* gain_control) const;
void AnalyzePreProcess(const AudioBuffer* audio);
void Process(const AudioBuffer* audio);
// Call when the capture stream has been muted/unmuted. This causes the
// manager to disregard all incoming audio; chances are good it's background
// noise to which we'd like to avoid adapting.
void SetCaptureMuted(bool muted);
bool capture_muted() { return capture_muted_; }
float voice_probability() const;
float voice_probability();
int stream_analog_level() const { return stream_analog_level_; }
void set_stream_analog_level(int level);
int num_channels() const { return num_capture_channels_; }
int sample_rate_hz() const { return sample_rate_hz_; }
// If available, returns a new compression gain for the digital gain control.
absl::optional<int> GetDigitalComressionGain();
private:
friend class AgcManagerDirectTest;
FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
DisableDigitalDisablesDigital);
FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
AgcMinMicLevelExperiment);
// Dependency injection for testing. Don't delete |agc| as the memory is owned
// by the manager.
AgcManagerDirect(Agc* agc,
int startup_min_level,
int clipped_level_min,
int sample_rate_hz);
void AnalyzePreProcess(const float* const* audio, size_t samples_per_channel);
void AggregateChannelLevels();
std::unique_ptr<ApmDataDumper> data_dumper_;
static int instance_counter_;
const bool use_min_channel_level_;
const int sample_rate_hz_;
const int num_capture_channels_;
const bool disable_digital_adaptive_;
int frames_since_clipped_;
int stream_analog_level_ = 0;
bool capture_muted_;
int channel_controlling_gain_ = 0;
std::vector<std::unique_ptr<MonoAgc>> channel_agcs_;
std::vector<absl::optional<int>> new_compressions_to_set_;
};
class MonoAgc {
public:
MonoAgc(ApmDataDumper* data_dumper,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int min_mic_level);
~MonoAgc();
MonoAgc(const MonoAgc&) = delete;
MonoAgc& operator=(const MonoAgc&) = delete;
void Initialize();
void SetCaptureMuted(bool muted);
void HandleClipping();
void Process(const int16_t* audio,
size_t samples_per_channel,
int sample_rate_hz);
void set_stream_analog_level(int level) { stream_analog_level_ = level; }
int stream_analog_level() const { return stream_analog_level_; }
float voice_probability() const { return agc_->voice_probability(); }
void ActivateLogging() { log_to_histograms_ = true; }
absl::optional<int> new_compression() const {
return new_compression_to_set_;
}
// Only used for testing.
void set_agc(Agc* agc) { agc_.reset(agc); }
int min_mic_level() const { return min_mic_level_; }
int startup_min_level() const { return startup_min_level_; }
private:
// Sets a new microphone level, after first checking that it hasn't been
@ -81,28 +149,26 @@ class AgcManagerDirect final {
void UpdateGain();
void UpdateCompressor();
rtc::scoped_ptr<Agc> agc_;
GainControl* gctrl_;
VolumeCallbacks* volume_callbacks_;
int frames_since_clipped_;
int level_;
const int min_mic_level_;
const bool disable_digital_adaptive_;
std::unique_ptr<Agc> agc_;
int level_ = 0;
int max_level_;
int max_compression_gain_;
int target_compression_;
int compression_;
float compression_accumulator_;
bool capture_muted_;
bool check_volume_on_next_process_;
bool startup_;
bool capture_muted_ = false;
bool check_volume_on_next_process_ = true;
bool startup_ = true;
int startup_min_level_;
rtc::scoped_ptr<DebugFile> file_preproc_;
rtc::scoped_ptr<DebugFile> file_postproc_;
RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
int calls_since_last_gain_log_ = 0;
int stream_analog_level_ = 0;
absl::optional<int> new_compression_to_set_;
bool log_to_histograms_ = false;
const int clipped_level_min_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#endif // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_