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.
AlgorithmicDelaySeconds(int source_rate_hz)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;
get_resampler_for_testing()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