cf808d2366
This change instroduces a mode where the Accelerate operation will be more aggressive. When enabled, it will allow acceleration at lower correlation levels, and possibly remove multiple pitch periods at once. The feature is enabled through NetEq::Config, and is off by default. This means that bit-exactness tests are currently not affected. A unit test was added for the Accelerate class, with and without fast mode enabled. BUG=4691 R=minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50039004 Cr-Commit-Position: refs/heads/master@{#9295}
102 lines
4.0 KiB
C++
102 lines
4.0 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/accelerate.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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namespace webrtc {
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Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
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size_t input_length,
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bool fast_accelerate,
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AudioMultiVector* output,
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int16_t* length_change_samples) {
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// Input length must be (almost) 30 ms.
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static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
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if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ <
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(2 * k15ms - 1) * fs_mult_) {
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// Length of input data too short to do accelerate. Simply move all data
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// from input to output.
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output->PushBackInterleaved(input, input_length);
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return kError;
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}
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return TimeStretch::Process(input, input_length, fast_accelerate, output,
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length_change_samples);
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}
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void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
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int16_t* best_correlation,
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int* /*peak_index*/) const {
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// When the signal does not contain any active speech, the correlation does
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// not matter. Simply set it to zero.
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*best_correlation = 0;
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}
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Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
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const int16_t* input,
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size_t input_length,
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size_t peak_index,
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int16_t best_correlation,
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bool active_speech,
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bool fast_mode,
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AudioMultiVector* output) const {
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// Check for strong correlation or passive speech.
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// Use 8192 (0.5 in Q14) in fast mode.
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const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
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if ((best_correlation > correlation_threshold) || !active_speech) {
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// Do accelerate operation by overlap add.
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// Pre-calculate common multiplication with |fs_mult_|.
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// 120 corresponds to 15 ms.
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size_t fs_mult_120 = fs_mult_ * 120;
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if (fast_mode) {
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// Fit as many multiples of |peak_index| as possible in fs_mult_120.
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// TODO(henrik.lundin) Consider finding multiple correlation peaks and
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// pick the one with the longest correlation lag in this case.
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peak_index = (fs_mult_120 / peak_index) * peak_index;
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}
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assert(fs_mult_120 >= peak_index); // Should be handled in Process().
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// Copy first part; 0 to 15 ms.
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output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
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// Copy the |peak_index| starting at 15 ms to |temp_vector|.
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AudioMultiVector temp_vector(num_channels_);
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temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
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peak_index * num_channels_);
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// Cross-fade |temp_vector| onto the end of |output|.
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output->CrossFade(temp_vector, peak_index);
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// Copy the last unmodified part, 15 ms + pitch period until the end.
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output->PushBackInterleaved(
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&input[(fs_mult_120 + peak_index) * num_channels_],
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input_length - (fs_mult_120 + peak_index) * num_channels_);
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if (active_speech) {
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return kSuccess;
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} else {
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return kSuccessLowEnergy;
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}
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} else {
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// Accelerate not allowed. Simply move all data from decoded to outData.
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output->PushBackInterleaved(input, input_length);
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return kNoStretch;
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}
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}
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Accelerate* AccelerateFactory::Create(
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int sample_rate_hz,
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size_t num_channels,
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const BackgroundNoise& background_noise) const {
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return new Accelerate(sample_rate_hz, num_channels, background_noise);
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}
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} // namespace webrtc
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