From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../modules/audio_coding/neteq/accelerate.cc       | 105 +++++++++++++++++++++
 1 file changed, 105 insertions(+)
 create mode 100644 third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc

(limited to 'third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc')

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
new file mode 100644
index 0000000000..06a38cc534
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
@@ -0,0 +1,105 @@
+/*
+ *  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.
+ */
+
+#include "modules/audio_coding/neteq/accelerate.h"
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
+                                            size_t input_length,
+                                            bool fast_accelerate,
+                                            AudioMultiVector* output,
+                                            size_t* length_change_samples) {
+  // Input length must be (almost) 30 ms.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
+    // Length of input data too short to do accelerate. Simply move all data
+    // from input to output.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kError;
+  }
+  return TimeStretch::Process(input, input_length, fast_accelerate, output,
+                              length_change_samples);
+}
+
+void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
+                                               int16_t* best_correlation,
+                                               size_t* /*peak_index*/) const {
+  // When the signal does not contain any active speech, the correlation does
+  // not matter. Simply set it to zero.
+  *best_correlation = 0;
+}
+
+Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool fast_mode,
+    AudioMultiVector* output) const {
+  // Check for strong correlation or passive speech.
+  // Use 8192 (0.5 in Q14) in fast mode.
+  const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
+  if ((best_correlation > correlation_threshold) || !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Pre-calculate common multiplication with `fs_mult_`.
+    // 120 corresponds to 15 ms.
+    size_t fs_mult_120 = fs_mult_ * 120;
+
+    if (fast_mode) {
+      // Fit as many multiples of `peak_index` as possible in fs_mult_120.
+      // TODO(henrik.lundin) Consider finding multiple correlation peaks and
+      // pick the one with the longest correlation lag in this case.
+      peak_index = (fs_mult_120 / peak_index) * peak_index;
+    }
+
+    RTC_DCHECK_GE(fs_mult_120, peak_index);  // Should be handled in Process().
+    // Copy first part; 0 to 15 ms.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
+    // Copy the `peak_index` starting at 15 ms to `temp_vector`.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[fs_mult_120 * num_channels_], peak_index * num_channels_));
+    // Cross-fade `temp_vector` onto the end of `output`.
+    output->CrossFade(temp_vector, peak_index);
+    // Copy the last unmodified part, 15 ms + pitch period until the end.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[(fs_mult_120 + peak_index) * num_channels_],
+        input_length - (fs_mult_120 + peak_index) * num_channels_));
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kNoStretch;
+  }
+}
+
+Accelerate* AccelerateFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise) const {
+  return new Accelerate(sample_rate_hz, num_channels, background_noise);
+}
+
+}  // namespace webrtc
-- 
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