Rename neteq4 folder to neteq

Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/preemptive_expand.cc b/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
new file mode 100644
index 0000000..b2dc3e6
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
@@ -0,0 +1,110 @@
+/*
+ *  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 "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
+
+#include <algorithm>  // min, max
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
+    const int16_t* input,
+    int input_length,
+    int old_data_length,
+    AudioMultiVector* output,
+    int16_t* length_change_samples) {
+  old_data_length_per_channel_ = old_data_length;
+  // Input length must be (almost) 30 ms.
+  // Also, the new part must be at least |overlap_samples_| elements.
+  static const int k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
+      old_data_length >= input_length / num_channels_ - overlap_samples_) {
+    // Length of input data too short to do preemptive expand. Simply move all
+    // data from input to output.
+    output->PushBackInterleaved(input, input_length);
+    return kError;
+  }
+  return TimeStretch::Process(input, input_length, output,
+                              length_change_samples);
+}
+
+void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
+                                                     int16_t* best_correlation,
+                                                     int* 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;
+
+  // For low energy expansion, the new data can be less than 15 ms,
+  // but we must ensure that best_correlation is not larger than the length of
+  // the new data.
+  // but we must ensure that best_correlation is not larger than the new data.
+  *peak_index = std::min(*peak_index,
+                         static_cast<int>(len - old_data_length_per_channel_));
+}
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
+    const int16_t *input, size_t input_length, size_t peak_index,
+    int16_t best_correlation, bool active_speech,
+    AudioMultiVector* output) const {
+  // Pre-calculate common multiplication with |fs_mult_|.
+  // 120 corresponds to 15 ms.
+  int fs_mult_120 = fs_mult_ * 120;
+  assert(old_data_length_per_channel_ >= 0);  // Make sure it's been set.
+  // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
+  // or passive speech.
+  if (((best_correlation > kCorrelationThreshold) &&
+      (old_data_length_per_channel_ <= fs_mult_120)) ||
+      !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Set length of the first part, not to be modified.
+    size_t unmodified_length = std::max(old_data_length_per_channel_,
+                                        fs_mult_120);
+    // Copy first part, including cross-fade region.
+    output->PushBackInterleaved(
+        input, (unmodified_length + peak_index) * num_channels_);
+    // Copy the last |peak_index| samples up to 15 ms to |temp_vector|.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(
+        &input[(unmodified_length - peak_index) * 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(
+        &input[unmodified_length * num_channels_],
+        input_length - unmodified_length * num_channels_);
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(input, input_length);
+    return kNoStretch;
+  }
+}
+
+PreemptiveExpand* PreemptiveExpandFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise,
+    int overlap_samples) const {
+  return new PreemptiveExpand(
+      sample_rate_hz, num_channels, background_noise, overlap_samples);
+}
+
+}  // namespace webrtc