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/*
* 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_MERGE_H_
#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
#include "modules/audio_coding/neteq/audio_multi_vector.h"
namespace webrtc {
// Forward declarations.
class Expand;
class SyncBuffer;
// This class handles the transition from expansion to normal operation.
// When a packet is not available for decoding when needed, the expand operation
// is called to generate extrapolation data. If the missing packet arrives,
// i.e., it was just delayed, it can be decoded and appended directly to the
// end of the expanded data (thanks to how the Expand class operates). However,
// if a later packet arrives instead, the loss is a fact, and the new data must
// be stitched together with the end of the expanded data. This stitching is
// what the Merge class does.
class Merge {
public:
Merge(int fs_hz,
size_t num_channels,
Expand* expand,
SyncBuffer* sync_buffer);
virtual ~Merge();
Merge(const Merge&) = delete;
Merge& operator=(const Merge&) = delete;
// The main method to produce the audio data. The decoded data is supplied in
// `input`, having `input_length` samples in total for all channels
// (interleaved). The result is written to `output`. The number of channels
// allocated in `output` defines the number of channels that will be used when
// de-interleaving `input`.
virtual size_t Process(int16_t* input,
size_t input_length,
AudioMultiVector* output);
virtual size_t RequiredFutureSamples();
protected:
const int fs_hz_;
const size_t num_channels_;
private:
static const int kMaxSampleRate = 48000;
static const size_t kExpandDownsampLength = 100;
static const size_t kInputDownsampLength = 40;
static const size_t kMaxCorrelationLength = 60;
// Calls `expand_` to get more expansion data to merge with. The data is
// written to `expanded_signal_`. Returns the length of the expanded data,
// while `expand_period` will be the number of samples in one expansion period
// (typically one pitch period). The value of `old_length` will be the number
// of samples that were taken from the `sync_buffer_`.
size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
// Analyzes `input` and `expanded_signal` and returns muting factor (Q14) to
// be used on the new data.
int16_t SignalScaling(const int16_t* input,
size_t input_length,
const int16_t* expanded_signal) const;
// Downsamples `input` (`input_length` samples) and `expanded_signal` to
// 4 kHz sample rate. The downsampled signals are written to
// `input_downsampled_` and `expanded_downsampled_`, respectively.
void Downsample(const int16_t* input,
size_t input_length,
const int16_t* expanded_signal,
size_t expanded_length);
// Calculates cross-correlation between `input_downsampled_` and
// `expanded_downsampled_`, and finds the correlation maximum. The maximizing
// lag is returned.
size_t CorrelateAndPeakSearch(size_t start_position,
size_t input_length,
size_t expand_period) const;
const int fs_mult_; // fs_hz_ / 8000.
const size_t timestamps_per_call_;
Expand* expand_;
SyncBuffer* sync_buffer_;
int16_t expanded_downsampled_[kExpandDownsampLength];
int16_t input_downsampled_[kInputDownsampLength];
AudioMultiVector expanded_;
std::vector<int16_t> temp_data_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MERGE_H_
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