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henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001/*
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
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#include "modules/audio_coding/neteq/preemptive_expand.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000012
13#include <algorithm> // min, max
14
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020015#include "common_audio/signal_processing/include/signal_processing_library.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000016
17namespace webrtc {
18
19PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
20 const int16_t* input,
Peter Kastingdce40cf2015-08-24 14:52:23 -070021 size_t input_length,
22 size_t old_data_length,
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000023 AudioMultiVector* output,
Peter Kastingdce40cf2015-08-24 14:52:23 -070024 size_t* length_change_samples) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000025 old_data_length_per_channel_ = old_data_length;
26 // Input length must be (almost) 30 ms.
27 // Also, the new part must be at least |overlap_samples_| elements.
Peter Kastingdce40cf2015-08-24 14:52:23 -070028 static const size_t k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000029 if (num_channels_ == 0 ||
30 input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
31 old_data_length >= input_length / num_channels_ - overlap_samples_) {
32 // Length of input data too short to do preemptive expand. Simply move all
33 // data from input to output.
34 output->PushBackInterleaved(input, input_length);
35 return kError;
36 }
Henrik Lundincf808d22015-05-27 14:33:29 +020037 const bool kFastMode = false; // Fast mode is not available for PE Expand.
38 return TimeStretch::Process(input, input_length, kFastMode, output,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000039 length_change_samples);
40}
41
turaj@webrtc.org362a55e2013-09-20 16:25:28 +000042void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000043 int16_t* best_correlation,
Peter Kastingdce40cf2015-08-24 14:52:23 -070044 size_t* peak_index) const {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000045 // When the signal does not contain any active speech, the correlation does
46 // not matter. Simply set it to zero.
47 *best_correlation = 0;
48
49 // For low energy expansion, the new data can be less than 15 ms,
50 // but we must ensure that best_correlation is not larger than the length of
51 // the new data.
52 // but we must ensure that best_correlation is not larger than the new data.
Yves Gerey665174f2018-06-19 15:03:05 +020053 *peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000054}
55
56PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
Henrik Lundincf808d22015-05-27 14:33:29 +020057 const int16_t* input,
58 size_t input_length,
59 size_t peak_index,
60 int16_t best_correlation,
61 bool active_speech,
62 bool /*fast_mode*/,
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000063 AudioMultiVector* output) const {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000064 // Pre-calculate common multiplication with |fs_mult_|.
65 // 120 corresponds to 15 ms.
Peter Kastingdce40cf2015-08-24 14:52:23 -070066 size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000067 // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
68 // or passive speech.
69 if (((best_correlation > kCorrelationThreshold) &&
Yves Gerey665174f2018-06-19 15:03:05 +020070 (old_data_length_per_channel_ <= fs_mult_120)) ||
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000071 !active_speech) {
72 // Do accelerate operation by overlap add.
73
74 // Set length of the first part, not to be modified.
Yves Gerey665174f2018-06-19 15:03:05 +020075 size_t unmodified_length =
76 std::max(old_data_length_per_channel_, fs_mult_120);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000077 // Copy first part, including cross-fade region.
78 output->PushBackInterleaved(
79 input, (unmodified_length + peak_index) * num_channels_);
80 // Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
henrik.lundin@webrtc.orgfd11bbf2013-09-30 20:38:44 +000081 AudioMultiVector temp_vector(num_channels_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000082 temp_vector.PushBackInterleaved(
83 &input[(unmodified_length - peak_index) * num_channels_],
84 peak_index * num_channels_);
85 // Cross-fade |temp_vector| onto the end of |output|.
86 output->CrossFade(temp_vector, peak_index);
87 // Copy the last unmodified part, 15 ms + pitch period until the end.
88 output->PushBackInterleaved(
89 &input[unmodified_length * num_channels_],
90 input_length - unmodified_length * num_channels_);
91
92 if (active_speech) {
93 return kSuccess;
94 } else {
95 return kSuccessLowEnergy;
96 }
97 } else {
98 // Accelerate not allowed. Simply move all data from decoded to outData.
99 output->PushBackInterleaved(input, input_length);
100 return kNoStretch;
101 }
102}
103
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +0000104PreemptiveExpand* PreemptiveExpandFactory::Create(
105 int sample_rate_hz,
106 size_t num_channels,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000107 const BackgroundNoise& background_noise,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700108 size_t overlap_samples) const {
Yves Gerey665174f2018-06-19 15:03:05 +0200109 return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise,
110 overlap_samples);
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +0000111}
112
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000113} // namespace webrtc