123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101 |
- /*
- * 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"
- #include "rtc_base/constructor_magic.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();
- // 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_;
- RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
- };
- } // namespace webrtc
- #endif // MODULES_AUDIO_CODING_NETEQ_MERGE_H_
|