mirror of
https://github.com/oxen-io/session-android.git
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191 lines
8.0 KiB
C++
191 lines
8.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq/normal.h"
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#include <string.h> // memset, memcpy
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#include <algorithm> // min
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
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#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
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#include "webrtc/modules/audio_coding/neteq/background_noise.h"
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#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
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#include "webrtc/modules/audio_coding/neteq/expand.h"
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#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
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namespace webrtc {
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int Normal::Process(const int16_t* input,
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size_t length,
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Modes last_mode,
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int16_t* external_mute_factor_array,
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AudioMultiVector* output) {
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if (length == 0) {
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// Nothing to process.
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output->Clear();
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return static_cast<int>(length);
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}
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assert(output->Empty());
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// Output should be empty at this point.
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output->PushBackInterleaved(input, length);
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int16_t* signal = &(*output)[0][0];
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const unsigned fs_mult = fs_hz_ / 8000;
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assert(fs_mult > 0);
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// fs_shift = log2(fs_mult), rounded down.
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// Note that |fs_shift| is not "exact" for 48 kHz.
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// TODO(hlundin): Investigate this further.
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const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
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// Check if last RecOut call resulted in an Expand. If so, we have to take
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// care of some cross-fading and unmuting.
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if (last_mode == kModeExpand) {
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// Generate interpolation data using Expand.
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// First, set Expand parameters to appropriate values.
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expand_->SetParametersForNormalAfterExpand();
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// Call Expand.
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AudioMultiVector expanded(output->Channels());
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expand_->Process(&expanded);
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expand_->Reset();
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for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
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// Adjust muting factor (main muting factor times expand muting factor).
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external_mute_factor_array[channel_ix] = static_cast<int16_t>(
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WEBRTC_SPL_MUL_16_16_RSFT(external_mute_factor_array[channel_ix],
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expand_->MuteFactor(channel_ix), 14));
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int16_t* signal = &(*output)[channel_ix][0];
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size_t length_per_channel = length / output->Channels();
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// Find largest absolute value in new data.
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int16_t decoded_max = WebRtcSpl_MaxAbsValueW16(
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signal, static_cast<int>(length_per_channel));
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// Adjust muting factor if needed (to BGN level).
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int energy_length = std::min(static_cast<int>(fs_mult * 64),
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static_cast<int>(length_per_channel));
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int scaling = 6 + fs_shift
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- WebRtcSpl_NormW32(decoded_max * decoded_max);
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scaling = std::max(scaling, 0); // |scaling| should always be >= 0.
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int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
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energy_length, scaling);
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energy = energy / (energy_length >> scaling);
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int mute_factor;
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if ((energy != 0) &&
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(energy > background_noise_.Energy(channel_ix))) {
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// Normalize new frame energy to 15 bits.
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scaling = WebRtcSpl_NormW32(energy) - 16;
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// We want background_noise_.energy() / energy in Q14.
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int32_t bgn_energy =
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background_noise_.Energy(channel_ix) << (scaling+14);
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int16_t energy_scaled = energy << scaling;
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int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
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mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
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} else {
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mute_factor = 16384; // 1.0 in Q14.
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}
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if (mute_factor > external_mute_factor_array[channel_ix]) {
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external_mute_factor_array[channel_ix] = std::min(mute_factor, 16384);
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}
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// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
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int16_t increment = 64 / fs_mult;
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for (size_t i = 0; i < length_per_channel; i++) {
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// Scale with mute factor.
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assert(channel_ix < output->Channels());
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assert(i < output->Size());
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int32_t scaled_signal = (*output)[channel_ix][i] *
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external_mute_factor_array[channel_ix];
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// Shift 14 with proper rounding.
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(*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
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// Increase mute_factor towards 16384.
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external_mute_factor_array[channel_ix] =
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std::min(external_mute_factor_array[channel_ix] + increment, 16384);
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}
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// Interpolate the expanded data into the new vector.
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// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
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assert(fs_shift < 3); // Will always be 0, 1, or, 2.
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increment = 4 >> fs_shift;
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int fraction = increment;
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for (size_t i = 0; i < 8 * fs_mult; i++) {
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// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
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// now for legacy bit-exactness.
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assert(channel_ix < output->Channels());
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assert(i < output->Size());
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(*output)[channel_ix][i] =
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(fraction * (*output)[channel_ix][i] +
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(32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
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fraction += increment;
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}
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}
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} else if (last_mode == kModeRfc3389Cng) {
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assert(output->Channels() == 1); // Not adapted for multi-channel yet.
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static const int kCngLength = 32;
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int16_t cng_output[kCngLength];
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// Reset mute factor and start up fresh.
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external_mute_factor_array[0] = 16384;
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AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
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if (cng_decoder) {
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CNG_dec_inst* cng_inst = static_cast<CNG_dec_inst*>(cng_decoder->state());
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// Generate long enough for 32kHz.
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if (WebRtcCng_Generate(cng_inst, cng_output, kCngLength, 0) < 0) {
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// Error returned; set return vector to all zeros.
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memset(cng_output, 0, sizeof(cng_output));
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}
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} else {
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// If no CNG instance is defined, just copy from the decoded data.
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// (This will result in interpolating the decoded with itself.)
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memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
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}
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// Interpolate the CNG into the new vector.
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// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
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assert(fs_shift < 3); // Will always be 0, 1, or, 2.
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int16_t increment = 4 >> fs_shift;
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int16_t fraction = increment;
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for (size_t i = 0; i < 8 * fs_mult; i++) {
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// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
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// for legacy bit-exactness.
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signal[i] =
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(fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
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fraction += increment;
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}
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} else if (external_mute_factor_array[0] < 16384) {
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// Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
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// still ramping up from previous muting.
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// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
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int16_t increment = 64 / fs_mult;
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size_t length_per_channel = length / output->Channels();
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for (size_t i = 0; i < length_per_channel; i++) {
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for (size_t channel_ix = 0; channel_ix < output->Channels();
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++channel_ix) {
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// Scale with mute factor.
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assert(channel_ix < output->Channels());
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assert(i < output->Size());
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int32_t scaled_signal = (*output)[channel_ix][i] *
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external_mute_factor_array[channel_ix];
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// Shift 14 with proper rounding.
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(*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
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// Increase mute_factor towards 16384.
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external_mute_factor_array[channel_ix] =
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std::min(16384, external_mute_factor_array[channel_ix] + increment);
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}
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}
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}
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return static_cast<int>(length);
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}
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} // namespace webrtc
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