189 lines
7.4 KiB
C++
189 lines
7.4 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 "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 "api/audio_codecs/audio_decoder.h"
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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#include "modules/audio_coding/neteq/audio_multi_vector.h"
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#include "modules/audio_coding/neteq/background_noise.h"
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#include "modules/audio_coding/neteq/decoder_database.h"
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#include "modules/audio_coding/neteq/expand.h"
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#include "rtc_base/checks.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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NetEq::Mode last_mode,
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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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RTC_DCHECK(output->Empty());
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// Output should be empty at this point.
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if (length % output->Channels() != 0) {
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// The length does not match the number of channels.
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output->Clear();
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return 0;
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}
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output->PushBackInterleaved(rtc::ArrayView<const int16_t>(input, length));
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const int fs_mult = fs_hz_ / 8000;
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RTC_DCHECK_GT(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 == NetEq::Mode::kExpand) {
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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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size_t length_per_channel = length / output->Channels();
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std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
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for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
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// Set muting factor to the same as expand muting factor.
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int16_t mute_factor = expand_->MuteFactor(channel_ix);
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(*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
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// Find largest absolute value in new data.
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int16_t decoded_max =
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WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
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// Adjust muting factor if needed (to BGN level).
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size_t energy_length =
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std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
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int scaling = 6 + fs_shift - 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.get(), signal.get(),
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energy_length, scaling);
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int32_t scaled_energy_length =
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static_cast<int32_t>(energy_length >> scaling);
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if (scaled_energy_length > 0) {
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energy = energy / scaled_energy_length;
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} else {
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energy = 0;
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}
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int local_mute_factor = 16384; // 1.0 in Q14.
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if ((energy != 0) && (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 = WEBRTC_SPL_SHIFT_W32(
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background_noise_.Energy(channel_ix), scaling + 14);
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int16_t energy_scaled =
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static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
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int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
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local_mute_factor =
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std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
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}
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mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
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RTC_DCHECK_LE(mute_factor, 16384);
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RTC_DCHECK_GE(mute_factor, 0);
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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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// or as fast as it takes to come back to full gain within the frame
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// length.
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const int back_to_fullscale_inc =
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static_cast<int>((16384 - mute_factor) / length_per_channel);
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const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
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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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RTC_DCHECK_LT(channel_ix, output->Channels());
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RTC_DCHECK_LT(i, output->Size());
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int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
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// Shift 14 with proper rounding.
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(*output)[channel_ix][i] =
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static_cast<int16_t>((scaled_signal + 8192) >> 14);
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// Increase mute_factor towards 16384.
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mute_factor =
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static_cast<int16_t>(std::min(mute_factor + 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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size_t win_length = samples_per_ms_;
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int16_t win_slope_Q14 = default_win_slope_Q14_;
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RTC_DCHECK_LT(channel_ix, output->Channels());
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if (win_length > output->Size()) {
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win_length = output->Size();
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win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
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}
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int16_t win_up_Q14 = 0;
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for (size_t i = 0; i < win_length; i++) {
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win_up_Q14 += win_slope_Q14;
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(*output)[channel_ix][i] =
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(win_up_Q14 * (*output)[channel_ix][i] +
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((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
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14;
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}
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RTC_DCHECK_GT(win_up_Q14,
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(1 << 14) - 32); // Worst case rouding is a length of 34
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}
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} else if (last_mode == NetEq::Mode::kRfc3389Cng) {
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RTC_DCHECK_EQ(output->Channels(), 1); // Not adapted for multi-channel yet.
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static const size_t kCngLength = 48;
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RTC_DCHECK_LE(8 * fs_mult, kCngLength);
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int16_t cng_output[kCngLength];
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ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
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if (cng_decoder) {
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// Generate long enough for 48kHz.
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if (!cng_decoder->Generate(cng_output, 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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(*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
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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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size_t win_length = samples_per_ms_;
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int16_t win_slope_Q14 = default_win_slope_Q14_;
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if (win_length > kCngLength) {
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win_length = kCngLength;
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win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
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}
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int16_t win_up_Q14 = 0;
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for (size_t i = 0; i < win_length; i++) {
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win_up_Q14 += win_slope_Q14;
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(*output)[0][i] =
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(win_up_Q14 * (*output)[0][i] +
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((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
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14;
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}
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RTC_DCHECK_GT(win_up_Q14,
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(1 << 14) - 32); // Worst case rouding is a length of 34
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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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