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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 |
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32 | 32 |
33 Agc::Agc() | 33 Agc::Agc() |
34 : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)), | 34 : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)), |
35 target_level_dbfs_(kDefaultLevelDbfs), | 35 target_level_dbfs_(kDefaultLevelDbfs), |
36 histogram_(Histogram::Create(kNumAnalysisFrames)), | 36 histogram_(Histogram::Create(kNumAnalysisFrames)), |
37 inactive_histogram_(Histogram::Create()) { | 37 inactive_histogram_(Histogram::Create()) { |
38 } | 38 } |
39 | 39 |
40 Agc::~Agc() {} | 40 Agc::~Agc() {} |
41 | 41 |
42 float Agc::AnalyzePreproc(const int16_t* audio, int length) { | 42 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) { |
43 assert(length > 0); | 43 assert(length > 0); |
44 int num_clipped = 0; | 44 size_t num_clipped = 0; |
45 for (int i = 0; i < length; ++i) { | 45 for (size_t i = 0; i < length; ++i) { |
46 if (audio[i] == 32767 || audio[i] == -32768) | 46 if (audio[i] == 32767 || audio[i] == -32768) |
47 ++num_clipped; | 47 ++num_clipped; |
48 } | 48 } |
49 return 1.0f * num_clipped / length; | 49 return 1.0f * num_clipped / length; |
50 } | 50 } |
51 | 51 |
52 int Agc::Process(const int16_t* audio, int length, int sample_rate_hz) { | 52 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) { |
53 vad_.ProcessChunk(audio, length, sample_rate_hz); | 53 vad_.ProcessChunk(audio, length, sample_rate_hz); |
54 const std::vector<double>& rms = vad_.chunkwise_rms(); | 54 const std::vector<double>& rms = vad_.chunkwise_rms(); |
55 const std::vector<double>& probabilities = | 55 const std::vector<double>& probabilities = |
56 vad_.chunkwise_voice_probabilities(); | 56 vad_.chunkwise_voice_probabilities(); |
57 DCHECK_EQ(rms.size(), probabilities.size()); | 57 DCHECK_EQ(rms.size(), probabilities.size()); |
58 for (size_t i = 0; i < rms.size(); ++i) { | 58 for (size_t i = 0; i < rms.size(); ++i) { |
59 histogram_->Update(rms[i], probabilities[i]); | 59 histogram_->Update(rms[i], probabilities[i]); |
60 } | 60 } |
61 return 0; | 61 return 0; |
62 } | 62 } |
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92 // limits. The upper limit should be chosen such that the risk of clipping is | 92 // limits. The upper limit should be chosen such that the risk of clipping is |
93 // low. The lower limit should not result in a too quiet signal. | 93 // low. The lower limit should not result in a too quiet signal. |
94 if (level >= 0 || level <= -100) | 94 if (level >= 0 || level <= -100) |
95 return -1; | 95 return -1; |
96 target_level_dbfs_ = level; | 96 target_level_dbfs_ = level; |
97 target_level_loudness_ = Dbfs2Loudness(level); | 97 target_level_loudness_ = Dbfs2Loudness(level); |
98 return 0; | 98 return 0; |
99 } | 99 } |
100 | 100 |
101 } // namespace webrtc | 101 } // namespace webrtc |
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