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#include "webrtc/modules/audio_coding/neteq/accelerate.h"
12
13#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
14
15namespace webrtc {
16
17Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
18                                            size_t input_length,
19                                            bool fast_accelerate,
20                                            AudioMultiVector* output,
21                                            size_t* length_change_samples) {
22  // Input length must be (almost) 30 ms.
23  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
24  if (num_channels_ == 0 ||
25      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
26    // Length of input data too short to do accelerate. Simply move all data
27    // from input to output.
28    output->PushBackInterleaved(input, input_length);
29    return kError;
30  }
31  return TimeStretch::Process(input, input_length, fast_accelerate, output,
32                              length_change_samples);
33}
34
35void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
36                                               int16_t* best_correlation,
37                                               size_t* /*peak_index*/) const {
38  // When the signal does not contain any active speech, the correlation does
39  // not matter. Simply set it to zero.
40  *best_correlation = 0;
41}
42
43Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
44    const int16_t* input,
45    size_t input_length,
46    size_t peak_index,
47    int16_t best_correlation,
48    bool active_speech,
49    bool fast_mode,
50    AudioMultiVector* output) const {
51  // Check for strong correlation or passive speech.
52  // Use 8192 (0.5 in Q14) in fast mode.
53  const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
54  if ((best_correlation > correlation_threshold) || !active_speech) {
55    // Do accelerate operation by overlap add.
56
57    // Pre-calculate common multiplication with |fs_mult_|.
58    // 120 corresponds to 15 ms.
59    size_t fs_mult_120 = fs_mult_ * 120;
60
61    if (fast_mode) {
62      // Fit as many multiples of |peak_index| as possible in fs_mult_120.
63      // TODO(henrik.lundin) Consider finding multiple correlation peaks and
64      // pick the one with the longest correlation lag in this case.
65      peak_index = (fs_mult_120 / peak_index) * peak_index;
66    }
67
68    assert(fs_mult_120 >= peak_index);  // Should be handled in Process().
69    // Copy first part; 0 to 15 ms.
70    output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
71    // Copy the |peak_index| starting at 15 ms to |temp_vector|.
72    AudioMultiVector temp_vector(num_channels_);
73    temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
74                                    peak_index * num_channels_);
75    // Cross-fade |temp_vector| onto the end of |output|.
76    output->CrossFade(temp_vector, peak_index);
77    // Copy the last unmodified part, 15 ms + pitch period until the end.
78    output->PushBackInterleaved(
79        &input[(fs_mult_120 + peak_index) * num_channels_],
80        input_length - (fs_mult_120 + peak_index) * num_channels_);
81
82    if (active_speech) {
83      return kSuccess;
84    } else {
85      return kSuccessLowEnergy;
86    }
87  } else {
88    // Accelerate not allowed. Simply move all data from decoded to outData.
89    output->PushBackInterleaved(input, input_length);
90    return kNoStretch;
91  }
92}
93
94Accelerate* AccelerateFactory::Create(
95    int sample_rate_hz,
96    size_t num_channels,
97    const BackgroundNoise& background_noise) const {
98  return new Accelerate(sample_rate_hz, num_channels, background_noise);
99}
100
101}  // namespace webrtc
102