session-android/jni/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

111 lines
4.4 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include <algorithm> // min, max
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
const int16_t* input,
int input_length,
int old_data_length,
AudioMultiVector* output,
int16_t* length_change_samples) {
old_data_length_per_channel_ = old_data_length;
// Input length must be (almost) 30 ms.
// Also, the new part must be at least |overlap_samples_| elements.
static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
if (num_channels_ == 0 ||
input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
old_data_length >= input_length / num_channels_ - overlap_samples_) {
// Length of input data too short to do preemptive expand. Simply move all
// data from input to output.
output->PushBackInterleaved(input, input_length);
return kError;
}
return TimeStretch::Process(input, input_length, output,
length_change_samples);
}
void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
int16_t* best_correlation,
int* peak_index) const {
// When the signal does not contain any active speech, the correlation does
// not matter. Simply set it to zero.
*best_correlation = 0;
// For low energy expansion, the new data can be less than 15 ms,
// but we must ensure that best_correlation is not larger than the length of
// the new data.
// but we must ensure that best_correlation is not larger than the new data.
*peak_index = std::min(*peak_index,
static_cast<int>(len - old_data_length_per_channel_));
}
PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
const int16_t *input, size_t input_length, size_t peak_index,
int16_t best_correlation, bool active_speech,
AudioMultiVector* output) const {
// Pre-calculate common multiplication with |fs_mult_|.
// 120 corresponds to 15 ms.
int fs_mult_120 = fs_mult_ * 120;
assert(old_data_length_per_channel_ >= 0); // Make sure it's been set.
// Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
// or passive speech.
if (((best_correlation > kCorrelationThreshold) &&
(old_data_length_per_channel_ <= fs_mult_120)) ||
!active_speech) {
// Do accelerate operation by overlap add.
// Set length of the first part, not to be modified.
size_t unmodified_length = std::max(old_data_length_per_channel_,
fs_mult_120);
// Copy first part, including cross-fade region.
output->PushBackInterleaved(
input, (unmodified_length + peak_index) * num_channels_);
// Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
AudioMultiVector temp_vector(num_channels_);
temp_vector.PushBackInterleaved(
&input[(unmodified_length - peak_index) * num_channels_],
peak_index * num_channels_);
// Cross-fade |temp_vector| onto the end of |output|.
output->CrossFade(temp_vector, peak_index);
// Copy the last unmodified part, 15 ms + pitch period until the end.
output->PushBackInterleaved(
&input[unmodified_length * num_channels_],
input_length - unmodified_length * num_channels_);
if (active_speech) {
return kSuccess;
} else {
return kSuccessLowEnergy;
}
} else {
// Accelerate not allowed. Simply move all data from decoded to outData.
output->PushBackInterleaved(input, input_length);
return kNoStretch;
}
}
PreemptiveExpand* PreemptiveExpandFactory::Create(
int sample_rate_hz,
size_t num_channels,
const BackgroundNoise& background_noise,
int overlap_samples) const {
return new PreemptiveExpand(
sample_rate_hz, num_channels, background_noise, overlap_samples);
}
} // namespace webrtc