Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
      12             : #define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
      13             : 
      14             : #include "webrtc/base/array_view.h"
      15             : #include "webrtc/base/optional.h"
      16             : #include "webrtc/typedefs.h"
      17             : 
      18             : namespace webrtc {
      19             : 
      20             : // Computes the root mean square (RMS) level in dBFs (decibels from digital
      21             : // full-scale) of audio data. The computation follows RFC 6465:
      22             : // https://tools.ietf.org/html/rfc6465
      23             : // with the intent that it can provide the RTP audio level indication.
      24             : //
      25             : // The expected approach is to provide constant-sized chunks of audio to
      26             : // Analyze(). When enough chunks have been accumulated to form a packet, call
      27             : // Average() to get the audio level indicator for the RTP header.
      28           0 : class RmsLevel {
      29             :  public:
      30             :   struct Levels {
      31             :     int average;
      32             :     int peak;
      33             :   };
      34             : 
      35             :   static constexpr int kMinLevelDb = 127;
      36             : 
      37             :   RmsLevel();
      38             :   ~RmsLevel();
      39             : 
      40             :   // Can be called to reset internal states, but is not required during normal
      41             :   // operation.
      42             :   void Reset();
      43             : 
      44             :   // Pass each chunk of audio to Analyze() to accumulate the level.
      45             :   void Analyze(rtc::ArrayView<const int16_t> data);
      46             : 
      47             :   // If all samples with the given |length| have a magnitude of zero, this is
      48             :   // a shortcut to avoid some computation.
      49             :   void AnalyzeMuted(size_t length);
      50             : 
      51             :   // Computes the RMS level over all data passed to Analyze() since the last
      52             :   // call to Average(). The returned value is positive but should be interpreted
      53             :   // as negative as per the RFC. It is constrained to [0, 127]. Resets the
      54             :   // internal state to start a new measurement period.
      55             :   int Average();
      56             : 
      57             :   // Like Average() above, but also returns the RMS peak value. Resets the
      58             :   // internal state to start a new measurement period.
      59             :   Levels AverageAndPeak();
      60             : 
      61             :  private:
      62             :   // Compares |block_size| with |block_size_|. If they are different, calls
      63             :   // Reset() and stores the new size.
      64             :   void CheckBlockSize(size_t block_size);
      65             : 
      66             :   float sum_square_;
      67             :   size_t sample_count_;
      68             :   float max_sum_square_;
      69             :   rtc::Optional<size_t> block_size_;
      70             : };
      71             : 
      72             : }  // namespace webrtc
      73             : 
      74             : #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
      75             :