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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96
webrtc/modules/audio_processing/gain_control_impl.h
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@ -8,75 +8,85 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <vector>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/processing_component.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc/gain_control.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
class CriticalSectionWrapper;
class GainControlImpl : public GainControl,
public ProcessingComponent {
class GainControlImpl : public GainControl {
public:
GainControlImpl(const AudioProcessing* apm,
CriticalSectionWrapper* crit);
virtual ~GainControlImpl();
GainControlImpl();
GainControlImpl(const GainControlImpl&) = delete;
GainControlImpl& operator=(const GainControlImpl&) = delete;
int ProcessRenderAudio(AudioBuffer* audio);
int AnalyzeCaptureAudio(AudioBuffer* audio);
int ProcessCaptureAudio(AudioBuffer* audio);
~GainControlImpl() override;
// ProcessingComponent implementation.
int Initialize() override;
void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
int AnalyzeCaptureAudio(const AudioBuffer& audio);
int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo);
void Initialize(size_t num_proc_channels, int sample_rate_hz);
static void PackRenderAudioBuffer(const AudioBuffer& audio,
std::vector<int16_t>* packed_buffer);
// GainControl implementation.
bool is_enabled() const override;
int stream_analog_level() override;
bool is_limiter_enabled() const override;
Mode mode() const override;
int stream_analog_level() const override;
bool is_limiter_enabled() const override { return limiter_enabled_; }
Mode mode() const override { return mode_; }
int set_mode(Mode mode) override;
int compression_gain_db() const override { return compression_gain_db_; }
int set_analog_level_limits(int minimum, int maximum) override;
int set_compression_gain_db(int gain) override;
int set_target_level_dbfs(int level) override;
int enable_limiter(bool enable) override;
int set_stream_analog_level(int level) override;
private:
struct MonoAgcState;
// GainControl implementation.
int Enable(bool enable) override;
int set_stream_analog_level(int level) override;
int set_mode(Mode mode) override;
int set_target_level_dbfs(int level) override;
int target_level_dbfs() const override;
int set_compression_gain_db(int gain) override;
int compression_gain_db() const override;
int enable_limiter(bool enable) override;
int set_analog_level_limits(int minimum, int maximum) override;
int analog_level_minimum() const override;
int analog_level_maximum() const override;
bool stream_is_saturated() const override;
int target_level_dbfs() const override { return target_level_dbfs_; }
int analog_level_minimum() const override { return minimum_capture_level_; }
int analog_level_maximum() const override { return maximum_capture_level_; }
bool stream_is_saturated() const override { return stream_is_saturated_; }
// ProcessingComponent implementation.
void* CreateHandle() const override;
int InitializeHandle(void* handle) const override;
int ConfigureHandle(void* handle) const override;
void DestroyHandle(void* handle) const override;
int num_handles_required() const override;
int GetHandleError(void* handle) const override;
int Configure();
const AudioProcessing* apm_;
CriticalSectionWrapper* crit_;
std::unique_ptr<ApmDataDumper> data_dumper_;
const bool use_legacy_gain_applier_;
Mode mode_;
int minimum_capture_level_;
int maximum_capture_level_;
bool limiter_enabled_;
int target_level_dbfs_;
int compression_gain_db_;
std::vector<int> capture_levels_;
int analog_capture_level_;
int analog_capture_level_ = 0;
bool was_analog_level_set_;
bool stream_is_saturated_;
std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_;
std::vector<int> capture_levels_;
absl::optional<size_t> num_proc_channels_;
absl::optional<int> sample_rate_hz_;
static int instance_counter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
#endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_