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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ |
13 | 13 |
14 #include <string.h> // Access to size_t. | 14 #include <string.h> // Access to size_t. |
15 | 15 |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
hlundin-webrtc
2017/03/27 13:29:06
#include "webrtc/base/checks.h"
(Because of Checke
| |
18 #include "webrtc/base/constructormagic.h" | 18 #include "webrtc/base/constructormagic.h" |
19 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" | 19 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" |
20 #include "webrtc/modules/audio_coding/neteq/defines.h" | 20 #include "webrtc/modules/audio_coding/neteq/defines.h" |
21 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 | 24 |
25 // Forward declarations. | 25 // Forward declarations. |
26 class BackgroundNoise; | 26 class BackgroundNoise; |
27 class DecoderDatabase; | 27 class DecoderDatabase; |
28 class Expand; | 28 class Expand; |
29 | 29 |
30 // This class provides the "Normal" DSP operation, that is performed when | 30 // This class provides the "Normal" DSP operation, that is performed when |
31 // there is no data loss, no need to stretch the timing of the signal, and | 31 // there is no data loss, no need to stretch the timing of the signal, and |
32 // no other "special circumstances" are at hand. | 32 // no other "special circumstances" are at hand. |
33 class Normal { | 33 class Normal { |
34 public: | 34 public: |
35 Normal(int fs_hz, DecoderDatabase* decoder_database, | 35 Normal(int fs_hz, |
36 DecoderDatabase* decoder_database, | |
36 const BackgroundNoise& background_noise, | 37 const BackgroundNoise& background_noise, |
37 Expand* expand) | 38 Expand* expand) |
38 : fs_hz_(fs_hz), | 39 : fs_hz_(fs_hz), |
39 decoder_database_(decoder_database), | 40 decoder_database_(decoder_database), |
40 background_noise_(background_noise), | 41 background_noise_(background_noise), |
41 expand_(expand) { | 42 expand_(expand), |
43 samples_per_ms_(fs_hz / 1000), | |
44 default_win_slope_Q14_((1 << 14) / samples_per_ms_) { | |
45 rtc::CheckedDivExact(fs_hz_, 1000); | |
hlundin-webrtc
2017/03/27 13:29:06
You should be able to do this where you perform an
| |
42 } | 46 } |
43 | 47 |
44 virtual ~Normal() {} | 48 virtual ~Normal() {} |
45 | 49 |
46 // Performs the "Normal" operation. The decoder data is supplied in |input|, | 50 // Performs the "Normal" operation. The decoder data is supplied in |input|, |
47 // having |length| samples in total for all channels (interleaved). The | 51 // having |length| samples in total for all channels (interleaved). The |
48 // result is written to |output|. The number of channels allocated in | 52 // result is written to |output|. The number of channels allocated in |
49 // |output| defines the number of channels that will be used when | 53 // |output| defines the number of channels that will be used when |
50 // de-interleaving |input|. |last_mode| contains the mode used in the previous | 54 // de-interleaving |input|. |last_mode| contains the mode used in the previous |
51 // GetAudio call (i.e., not the current one), and |external_mute_factor| is | 55 // GetAudio call (i.e., not the current one), and |external_mute_factor| is |
52 // a pointer to the mute factor in the NetEqImpl class. | 56 // a pointer to the mute factor in the NetEqImpl class. |
53 int Process(const int16_t* input, size_t length, | 57 int Process(const int16_t* input, size_t length, |
54 Modes last_mode, | 58 Modes last_mode, |
55 int16_t* external_mute_factor_array, | 59 int16_t* external_mute_factor_array, |
56 AudioMultiVector* output); | 60 AudioMultiVector* output); |
57 | 61 |
58 private: | 62 private: |
59 int fs_hz_; | 63 int fs_hz_; |
60 DecoderDatabase* decoder_database_; | 64 DecoderDatabase* decoder_database_; |
61 const BackgroundNoise& background_noise_; | 65 const BackgroundNoise& background_noise_; |
62 Expand* expand_; | 66 Expand* expand_; |
67 const size_t samples_per_ms_; | |
68 const int16_t default_win_slope_Q14_; | |
63 | 69 |
64 RTC_DISALLOW_COPY_AND_ASSIGN(Normal); | 70 RTC_DISALLOW_COPY_AND_ASSIGN(Normal); |
65 }; | 71 }; |
66 | 72 |
67 } // namespace webrtc | 73 } // namespace webrtc |
68 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ | 74 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ |
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