Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

115 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,
size_t input_length,
size_t old_data_length,
AudioMultiVector* output,
size_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 size_t 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;
}
const bool kFastMode = false; // Fast mode is not available for PE Expand.
return TimeStretch::Process(input, input_length, kFastMode, output,
length_change_samples);
}
void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
int16_t* best_correlation,
size_t* 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,
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,
bool /*fast_mode*/,
AudioMultiVector* output) const {
// Pre-calculate common multiplication with |fs_mult_|.
// 120 corresponds to 15 ms.
size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
// 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,
size_t overlap_samples) const {
return new PreemptiveExpand(
sample_rate_hz, num_channels, background_noise, overlap_samples);
}
} // namespace webrtc