2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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.
11 #include "webrtc/modules/audio_coding/neteq/normal.h"
13 #include <string.h> // memset, memcpy
15 #include <algorithm> // min
17 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
18 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
19 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
20 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
21 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
22 #include "webrtc/modules/audio_coding/neteq/expand.h"
23 #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
27 int Normal::Process(const int16_t* input,
30 int16_t* external_mute_factor_array,
31 AudioMultiVector* output) {
33 // Nothing to process.
35 return static_cast<int>(length);
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.
45 output->PushBackInterleaved(input, length);
46 int16_t* signal = &(*output)[0][0];
48 const unsigned fs_mult = fs_hz_ / 8000;
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);
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();
63 AudioMultiVector expanded(output->Channels());
64 expand_->Process(&expanded);
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 WEBRTC_SPL_MUL_16_16_RSFT(external_mute_factor_array[channel_ix],
71 expand_->MuteFactor(channel_ix), 14));
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 = WebRtcSpl_MaxAbsValueW16(
77 signal, static_cast<int>(length_per_channel));
78 // Adjust muting factor if needed (to BGN level).
79 int energy_length = std::min(static_cast<int>(fs_mult * 64),
80 static_cast<int>(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 if ((energy_length >> scaling) > 0) {
87 energy = energy / (energy_length >> scaling);
94 (energy > background_noise_.Energy(channel_ix))) {
95 // Normalize new frame energy to 15 bits.
96 scaling = WebRtcSpl_NormW32(energy) - 16;
97 // We want background_noise_.energy() / energy in Q14.
99 background_noise_.Energy(channel_ix) << (scaling+14);
100 int16_t energy_scaled = energy << scaling;
101 int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
102 mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
104 mute_factor = 16384; // 1.0 in Q14.
106 if (mute_factor > external_mute_factor_array[channel_ix]) {
107 external_mute_factor_array[channel_ix] = std::min(mute_factor, 16384);
110 // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
111 int16_t increment = 64 / fs_mult;
112 for (size_t i = 0; i < length_per_channel; i++) {
113 // Scale with mute factor.
114 assert(channel_ix < output->Channels());
115 assert(i < output->Size());
116 int32_t scaled_signal = (*output)[channel_ix][i] *
117 external_mute_factor_array[channel_ix];
118 // Shift 14 with proper rounding.
119 (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
120 // Increase mute_factor towards 16384.
121 external_mute_factor_array[channel_ix] =
122 std::min(external_mute_factor_array[channel_ix] + increment, 16384);
125 // Interpolate the expanded data into the new vector.
126 // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
127 assert(fs_shift < 3); // Will always be 0, 1, or, 2.
128 increment = 4 >> fs_shift;
129 int fraction = increment;
130 for (size_t i = 0; i < 8 * fs_mult; i++) {
131 // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
132 // now for legacy bit-exactness.
133 assert(channel_ix < output->Channels());
134 assert(i < output->Size());
135 (*output)[channel_ix][i] =
136 (fraction * (*output)[channel_ix][i] +
137 (32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
138 fraction += increment;
141 } else if (last_mode == kModeRfc3389Cng) {
142 assert(output->Channels() == 1); // Not adapted for multi-channel yet.
143 static const int kCngLength = 32;
144 int16_t cng_output[kCngLength];
145 // Reset mute factor and start up fresh.
146 external_mute_factor_array[0] = 16384;
147 AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
150 CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
151 // Generate long enough for 32kHz.
152 if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
153 // Error returned; set return vector to all zeros.
154 memset(cng_output, 0, sizeof(cng_output));
157 // If no CNG instance is defined, just copy from the decoded data.
158 // (This will result in interpolating the decoded with itself.)
159 memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
161 // Interpolate the CNG into the new vector.
162 // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
163 assert(fs_shift < 3); // Will always be 0, 1, or, 2.
164 int16_t increment = 4 >> fs_shift;
165 int16_t fraction = increment;
166 for (size_t i = 0; i < 8 * fs_mult; i++) {
167 // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
168 // for legacy bit-exactness.
170 (fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
171 fraction += increment;
173 } else if (external_mute_factor_array[0] < 16384) {
174 // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
175 // still ramping up from previous muting.
176 // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
177 int16_t increment = 64 / fs_mult;
178 size_t length_per_channel = length / output->Channels();
179 for (size_t i = 0; i < length_per_channel; i++) {
180 for (size_t channel_ix = 0; channel_ix < output->Channels();
182 // Scale with mute factor.
183 assert(channel_ix < output->Channels());
184 assert(i < output->Size());
185 int32_t scaled_signal = (*output)[channel_ix][i] *
186 external_mute_factor_array[channel_ix];
187 // Shift 14 with proper rounding.
188 (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
189 // Increase mute_factor towards 16384.
190 external_mute_factor_array[channel_ix] =
191 std::min(16384, external_mute_factor_array[channel_ix] + increment);
196 return static_cast<int>(length);
199 } // namespace webrtc