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      1 /*
      2  *  Copyright (c) 2013 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_RESAMPLER_PUSH_SINC_RESAMPLER_H_
     12 #define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
     13 
     14 #include "webrtc/base/constructormagic.h"
     15 #include "webrtc/base/scoped_ptr.h"
     16 #include "webrtc/common_audio/resampler/sinc_resampler.h"
     17 #include "webrtc/typedefs.h"
     18 
     19 namespace webrtc {
     20 
     21 // A thin wrapper over SincResampler to provide a push-based interface as
     22 // required by WebRTC. SincResampler uses a pull-based interface, and will
     23 // use SincResamplerCallback::Run() to request data upon a call to Resample().
     24 // These Run() calls will happen on the same thread Resample() is called on.
     25 class PushSincResampler : public SincResamplerCallback {
     26  public:
     27   // Provide the size of the source and destination blocks in samples. These
     28   // must correspond to the same time duration (typically 10 ms) as the sample
     29   // ratio is inferred from them.
     30   PushSincResampler(size_t source_frames, size_t destination_frames);
     31   ~PushSincResampler() override;
     32 
     33   // Perform the resampling. |source_frames| must always equal the
     34   // |source_frames| provided at construction. |destination_capacity| must be
     35   // at least as large as |destination_frames|. Returns the number of samples
     36   // provided in destination (for convenience, since this will always be equal
     37   // to |destination_frames|).
     38   size_t Resample(const int16_t* source, size_t source_frames,
     39                   int16_t* destination, size_t destination_capacity);
     40   size_t Resample(const float* source,
     41                   size_t source_frames,
     42                   float* destination,
     43                   size_t destination_capacity);
     44 
     45   // Delay due to the filter kernel. Essentially, the time after which an input
     46   // sample will appear in the resampled output.
     47   static float AlgorithmicDelaySeconds(int source_rate_hz) {
     48     return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
     49   }
     50 
     51  protected:
     52   // Implements SincResamplerCallback.
     53   void Run(size_t frames, float* destination) override;
     54 
     55  private:
     56   friend class PushSincResamplerTest;
     57   SincResampler* get_resampler_for_testing() { return resampler_.get(); }
     58 
     59   rtc::scoped_ptr<SincResampler> resampler_;
     60   rtc::scoped_ptr<float[]> float_buffer_;
     61   const float* source_ptr_;
     62   const int16_t* source_ptr_int_;
     63   const size_t destination_frames_;
     64 
     65   // True on the first call to Resample(), to prime the SincResampler buffer.
     66   bool first_pass_;
     67 
     68   // Used to assert we are only requested for as much data as is available.
     69   size_t source_available_;
     70 
     71   RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
     72 };
     73 
     74 }  // namespace webrtc
     75 
     76 #endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
     77