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/normal.h"
12
13 #include <string.h> // memset, memcpy
14
15 #include <algorithm> // min
16
17 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
18 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
19 #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
20 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
21 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
22 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
23 #include "webrtc/modules/audio_coding/neteq/expand.h"
24
25 namespace webrtc {
26
Process(const int16_t * input,size_t length,Modes last_mode,int16_t * external_mute_factor_array,AudioMultiVector * output)27 int Normal::Process(const int16_t* input,
28 size_t length,
29 Modes last_mode,
30 int16_t* external_mute_factor_array,
31 AudioMultiVector* output) {
32 if (length == 0) {
33 // Nothing to process.
34 output->Clear();
35 return static_cast<int>(length);
36 }
37
38 assert(output->Empty());
39 // Output should be empty at this point.
40 if (length % output->Channels() != 0) {
41 // The length does not match the number of channels.
42 output->Clear();
43 return 0;
44 }
45 output->PushBackInterleaved(input, length);
46 int16_t* signal = &(*output)[0][0];
47
48 const int fs_mult = fs_hz_ / 8000;
49 assert(fs_mult > 0);
50 // fs_shift = log2(fs_mult), rounded down.
51 // Note that |fs_shift| is not "exact" for 48 kHz.
52 // TODO(hlundin): Investigate this further.
53 const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
54
55 // Check if last RecOut call resulted in an Expand. If so, we have to take
56 // care of some cross-fading and unmuting.
57 if (last_mode == kModeExpand) {
58 // Generate interpolation data using Expand.
59 // First, set Expand parameters to appropriate values.
60 expand_->SetParametersForNormalAfterExpand();
61
62 // Call Expand.
63 AudioMultiVector expanded(output->Channels());
64 expand_->Process(&expanded);
65 expand_->Reset();
66
67 for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
68 // Adjust muting factor (main muting factor times expand muting factor).
69 external_mute_factor_array[channel_ix] = static_cast<int16_t>(
70 (external_mute_factor_array[channel_ix] *
71 expand_->MuteFactor(channel_ix)) >> 14);
72
73 int16_t* signal = &(*output)[channel_ix][0];
74 size_t length_per_channel = length / output->Channels();
75 // Find largest absolute value in new data.
76 int16_t decoded_max =
77 WebRtcSpl_MaxAbsValueW16(signal, length_per_channel);
78 // Adjust muting factor if needed (to BGN level).
79 size_t energy_length =
80 std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
81 int scaling = 6 + fs_shift
82 - WebRtcSpl_NormW32(decoded_max * decoded_max);
83 scaling = std::max(scaling, 0); // |scaling| should always be >= 0.
84 int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
85 energy_length, scaling);
86 int32_t scaled_energy_length =
87 static_cast<int32_t>(energy_length >> scaling);
88 if (scaled_energy_length > 0) {
89 energy = energy / scaled_energy_length;
90 } else {
91 energy = 0;
92 }
93
94 int mute_factor;
95 if ((energy != 0) &&
96 (energy > background_noise_.Energy(channel_ix))) {
97 // Normalize new frame energy to 15 bits.
98 scaling = WebRtcSpl_NormW32(energy) - 16;
99 // We want background_noise_.energy() / energy in Q14.
100 int32_t bgn_energy =
101 background_noise_.Energy(channel_ix) << (scaling+14);
102 int16_t energy_scaled = static_cast<int16_t>(energy << scaling);
103 int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
104 mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
105 } else {
106 mute_factor = 16384; // 1.0 in Q14.
107 }
108 if (mute_factor > external_mute_factor_array[channel_ix]) {
109 external_mute_factor_array[channel_ix] =
110 static_cast<int16_t>(std::min(mute_factor, 16384));
111 }
112
113 // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
114 int increment = 64 / fs_mult;
115 for (size_t i = 0; i < length_per_channel; i++) {
116 // Scale with mute factor.
117 assert(channel_ix < output->Channels());
118 assert(i < output->Size());
119 int32_t scaled_signal = (*output)[channel_ix][i] *
120 external_mute_factor_array[channel_ix];
121 // Shift 14 with proper rounding.
122 (*output)[channel_ix][i] =
123 static_cast<int16_t>((scaled_signal + 8192) >> 14);
124 // Increase mute_factor towards 16384.
125 external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
126 external_mute_factor_array[channel_ix] + increment, 16384));
127 }
128
129 // Interpolate the expanded data into the new vector.
130 // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
131 assert(fs_shift < 3); // Will always be 0, 1, or, 2.
132 increment = 4 >> fs_shift;
133 int fraction = increment;
134 for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
135 // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
136 // now for legacy bit-exactness.
137 assert(channel_ix < output->Channels());
138 assert(i < output->Size());
139 (*output)[channel_ix][i] =
140 static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
141 (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
142 fraction += increment;
143 }
144 }
145 } else if (last_mode == kModeRfc3389Cng) {
146 assert(output->Channels() == 1); // Not adapted for multi-channel yet.
147 static const size_t kCngLength = 32;
148 int16_t cng_output[kCngLength];
149 // Reset mute factor and start up fresh.
150 external_mute_factor_array[0] = 16384;
151 AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
152
153 if (cng_decoder) {
154 // Generate long enough for 32kHz.
155 if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
156 kCngLength, 0) < 0) {
157 // Error returned; set return vector to all zeros.
158 memset(cng_output, 0, sizeof(cng_output));
159 }
160 } else {
161 // If no CNG instance is defined, just copy from the decoded data.
162 // (This will result in interpolating the decoded with itself.)
163 memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
164 }
165 // Interpolate the CNG into the new vector.
166 // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
167 assert(fs_shift < 3); // Will always be 0, 1, or, 2.
168 int16_t increment = 4 >> fs_shift;
169 int16_t fraction = increment;
170 for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
171 // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
172 // for legacy bit-exactness.
173 signal[i] =
174 (fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
175 fraction += increment;
176 }
177 } else if (external_mute_factor_array[0] < 16384) {
178 // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
179 // still ramping up from previous muting.
180 // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
181 int increment = 64 / fs_mult;
182 size_t length_per_channel = length / output->Channels();
183 for (size_t i = 0; i < length_per_channel; i++) {
184 for (size_t channel_ix = 0; channel_ix < output->Channels();
185 ++channel_ix) {
186 // Scale with mute factor.
187 assert(channel_ix < output->Channels());
188 assert(i < output->Size());
189 int32_t scaled_signal = (*output)[channel_ix][i] *
190 external_mute_factor_array[channel_ix];
191 // Shift 14 with proper rounding.
192 (*output)[channel_ix][i] =
193 static_cast<int16_t>((scaled_signal + 8192) >> 14);
194 // Increase mute_factor towards 16384.
195 external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
196 16384, external_mute_factor_array[channel_ix] + increment));
197 }
198 }
199 }
200
201 return static_cast<int>(length);
202 }
203
204 } // namespace webrtc
205