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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/merge.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/merge.h | 101 |
1 files changed, 101 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge.h b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h new file mode 100644 index 0000000000..2f27106bfe --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h @@ -0,0 +1,101 @@ +/* + * 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_ |