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_COMMON_AUDIO_AUDIO_CONVERTER_H_
      12             : #define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
      13             : 
      14             : #include <memory>
      15             : 
      16             : #include "webrtc/base/constructormagic.h"
      17             : 
      18             : namespace webrtc {
      19             : 
      20             : // Format conversion (remixing and resampling) for audio. Only simple remixing
      21             : // conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
      22             : // upmix from mono (i.e. |src_channels == 1|).
      23             : //
      24             : // The source and destination chunks have the same duration in time; specifying
      25             : // the number of frames is equivalent to specifying the sample rates.
      26             : class AudioConverter {
      27             :  public:
      28             :   // Returns a new AudioConverter, which will use the supplied format for its
      29             :   // lifetime. Caller is responsible for the memory.
      30             :   static std::unique_ptr<AudioConverter> Create(size_t src_channels,
      31             :                                                 size_t src_frames,
      32             :                                                 size_t dst_channels,
      33             :                                                 size_t dst_frames);
      34           0 :   virtual ~AudioConverter() {};
      35             : 
      36             :   // Convert |src|, containing |src_size| samples, to |dst|, having a sample
      37             :   // capacity of |dst_capacity|. Both point to a series of buffers containing
      38             :   // the samples for each channel. The sizes must correspond to the format
      39             :   // passed to Create().
      40             :   virtual void Convert(const float* const* src, size_t src_size,
      41             :                        float* const* dst, size_t dst_capacity) = 0;
      42             : 
      43           0 :   size_t src_channels() const { return src_channels_; }
      44           0 :   size_t src_frames() const { return src_frames_; }
      45           0 :   size_t dst_channels() const { return dst_channels_; }
      46           0 :   size_t dst_frames() const { return dst_frames_; }
      47             : 
      48             :  protected:
      49             :   AudioConverter();
      50             :   AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
      51             :                  size_t dst_frames);
      52             : 
      53             :   // Helper to RTC_CHECK that inputs are correctly sized.
      54             :   void CheckSizes(size_t src_size, size_t dst_capacity) const;
      55             : 
      56             :  private:
      57             :   const size_t src_channels_;
      58             :   const size_t src_frames_;
      59             :   const size_t dst_channels_;
      60             :   const size_t dst_frames_;
      61             : 
      62             :   RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
      63             : };
      64             : 
      65             : }  // namespace webrtc
      66             : 
      67             : #endif  // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_