Line data Source code
1 : /*
2 : * Copyright (c) 2012 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_CODING_NETEQ_ACCELERATE_H_
12 : #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
13 :
14 : #include <assert.h>
15 :
16 : #include "webrtc/base/constructormagic.h"
17 : #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
18 : #include "webrtc/modules/audio_coding/neteq/time_stretch.h"
19 : #include "webrtc/typedefs.h"
20 :
21 : namespace webrtc {
22 :
23 : // Forward declarations.
24 : class BackgroundNoise;
25 :
26 : // This class implements the Accelerate operation. Most of the work is done
27 : // in the base class TimeStretch, which is shared with the PreemptiveExpand
28 : // operation. In the Accelerate class, the operations that are specific to
29 : // Accelerate are implemented.
30 0 : class Accelerate : public TimeStretch {
31 : public:
32 0 : Accelerate(int sample_rate_hz, size_t num_channels,
33 : const BackgroundNoise& background_noise)
34 0 : : TimeStretch(sample_rate_hz, num_channels, background_noise) {
35 0 : }
36 :
37 : // This method performs the actual Accelerate operation. The samples are
38 : // read from |input|, of length |input_length| elements, and are written to
39 : // |output|. The number of samples removed through time-stretching is
40 : // is provided in the output |length_change_samples|. The method returns
41 : // the outcome of the operation as an enumerator value. If |fast_accelerate|
42 : // is true, the algorithm will relax the requirements on finding strong
43 : // correlations, and may remove multiple pitch periods if possible.
44 : ReturnCodes Process(const int16_t* input,
45 : size_t input_length,
46 : bool fast_accelerate,
47 : AudioMultiVector* output,
48 : size_t* length_change_samples);
49 :
50 : protected:
51 : // Sets the parameters |best_correlation| and |peak_index| to suitable
52 : // values when the signal contains no active speech.
53 : void SetParametersForPassiveSpeech(size_t len,
54 : int16_t* best_correlation,
55 : size_t* peak_index) const override;
56 :
57 : // Checks the criteria for performing the time-stretching operation and,
58 : // if possible, performs the time-stretching.
59 : ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
60 : size_t input_length,
61 : size_t peak_index,
62 : int16_t best_correlation,
63 : bool active_speech,
64 : bool fast_mode,
65 : AudioMultiVector* output) const override;
66 :
67 : private:
68 : RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate);
69 : };
70 :
71 : struct AccelerateFactory {
72 0 : AccelerateFactory() {}
73 0 : virtual ~AccelerateFactory() {}
74 :
75 : virtual Accelerate* Create(int sample_rate_hz,
76 : size_t num_channels,
77 : const BackgroundNoise& background_noise) const;
78 : };
79 :
80 : } // namespace webrtc
81 : #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
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