webrtc/modules/audio_coding/neteq/expand.h
Mirko Bonadei 25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00

155 lines
4.7 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
#define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
#include <assert.h>
#include <memory>
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
// Forward declarations.
class AudioMultiVector;
class BackgroundNoise;
class RandomVector;
class StatisticsCalculator;
class SyncBuffer;
// This class handles extrapolation of audio data from the sync_buffer to
// produce packet-loss concealment.
// TODO(hlundin): Refactor this class to divide the long methods into shorter
// ones.
class Expand {
public:
Expand(BackgroundNoise* background_noise,
SyncBuffer* sync_buffer,
RandomVector* random_vector,
StatisticsCalculator* statistics,
int fs,
size_t num_channels);
virtual ~Expand();
// Resets the object.
virtual void Reset();
// The main method to produce concealment data. The data is appended to the
// end of |output|.
virtual int Process(AudioMultiVector* output);
// Prepare the object to do extra expansion during normal operation following
// a period of expands.
virtual void SetParametersForNormalAfterExpand();
// Prepare the object to do extra expansion during merge operation following
// a period of expands.
virtual void SetParametersForMergeAfterExpand();
// Returns the mute factor for |channel|.
int16_t MuteFactor(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
return channel_parameters_[channel].mute_factor;
}
// Returns true if expansion has been faded down to zero amplitude (for all
// channels); false otherwise.
bool Muted() const;
// Accessors and mutators.
virtual size_t overlap_length() const;
size_t max_lag() const { return max_lag_; }
protected:
static const int kMaxConsecutiveExpands = 200;
void GenerateRandomVector(int16_t seed_increment,
size_t length,
int16_t* random_vector);
// Initializes member variables at the beginning of an expand period.
void InitializeForAnExpandPeriod();
bool TooManyExpands();
// Analyzes the signal history in |sync_buffer_|, and set up all parameters
// necessary to produce concealment data.
void AnalyzeSignal(int16_t* random_vector);
RandomVector* const random_vector_;
SyncBuffer* const sync_buffer_;
bool first_expand_;
const int fs_hz_;
const size_t num_channels_;
int consecutive_expands_;
private:
static const size_t kUnvoicedLpcOrder = 6;
static const size_t kNumCorrelationCandidates = 3;
static const size_t kDistortionLength = 20;
static const size_t kLpcAnalysisLength = 160;
static const size_t kMaxSampleRate = 48000;
static const int kNumLags = 3;
struct ChannelParameters {
ChannelParameters();
int16_t mute_factor;
int16_t ar_filter[kUnvoicedLpcOrder + 1];
int16_t ar_filter_state[kUnvoicedLpcOrder];
int16_t ar_gain;
int16_t ar_gain_scale;
int16_t voice_mix_factor; /* Q14 */
int16_t current_voice_mix_factor; /* Q14 */
AudioVector expand_vector0;
AudioVector expand_vector1;
bool onset;
int mute_slope; /* Q20 */
};
// Calculate the auto-correlation of |input|, with length |input_length|
// samples. The correlation is calculated from a downsampled version of
// |input|, and is written to |output|.
void Correlation(const int16_t* input,
size_t input_length,
int16_t* output) const;
void UpdateLagIndex();
BackgroundNoise* const background_noise_;
StatisticsCalculator* const statistics_;
const size_t overlap_length_;
size_t max_lag_;
size_t expand_lags_[kNumLags];
int lag_index_direction_;
int current_lag_index_;
bool stop_muting_;
size_t expand_duration_samples_;
std::unique_ptr<ChannelParameters[]> channel_parameters_;
RTC_DISALLOW_COPY_AND_ASSIGN(Expand);
};
struct ExpandFactory {
ExpandFactory() {}
virtual ~ExpandFactory() {}
virtual Expand* Create(BackgroundNoise* background_noise,
SyncBuffer* sync_buffer,
RandomVector* random_vector,
StatisticsCalculator* statistics,
int fs,
size_t num_channels) const;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_EXPAND_H_