Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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_