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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56
webrtc/modules/audio_processing/rms_level.h
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@ -8,12 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#include <cstddef>
#include <stddef.h>
#include <stdint.h>
#include "webrtc/typedefs.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
namespace webrtc {
@ -23,37 +25,53 @@ namespace webrtc {
// with the intent that it can provide the RTP audio level indication.
//
// The expected approach is to provide constant-sized chunks of audio to
// Process(). When enough chunks have been accumulated to form a packet, call
// RMS() to get the audio level indicator for the RTP header.
class RMSLevel {
// Analyze(). When enough chunks have been accumulated to form a packet, call
// Average() to get the audio level indicator for the RTP header.
class RmsLevel {
public:
static const int kMinLevel = 127;
struct Levels {
int average;
int peak;
};
RMSLevel();
~RMSLevel();
enum : int { kMinLevelDb = 127 };
RmsLevel();
~RmsLevel();
// Can be called to reset internal states, but is not required during normal
// operation.
void Reset();
// Pass each chunk of audio to Process() to accumulate the level.
void Process(const int16_t* data, size_t length);
// Pass each chunk of audio to Analyze() to accumulate the level.
void Analyze(rtc::ArrayView<const int16_t> data);
void Analyze(rtc::ArrayView<const float> data);
// If all samples with the given |length| have a magnitude of zero, this is
// a shortcut to avoid some computation.
void ProcessMuted(size_t length);
void AnalyzeMuted(size_t length);
// Computes the RMS level over all data passed to Process() since the last
// call to RMS(). The returned value is positive but should be interpreted as
// negative as per the RFC. It is constrained to [0, 127].
int RMS();
// Computes the RMS level over all data passed to Analyze() since the last
// call to Average(). The returned value is positive but should be interpreted
// as negative as per the RFC. It is constrained to [0, 127]. Resets the
// internal state to start a new measurement period.
int Average();
// Like Average() above, but also returns the RMS peak value. Resets the
// internal state to start a new measurement period.
Levels AverageAndPeak();
private:
// Compares |block_size| with |block_size_|. If they are different, calls
// Reset() and stores the new size.
void CheckBlockSize(size_t block_size);
float sum_square_;
size_t sample_count_;
float max_sum_square_;
absl::optional<size_t> block_size_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_