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/accelerate.cc b/webrtc/modules/audio_coding/neteq/accelerate.cc
new file mode 100644
index 0000000..6acd778
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/accelerate.cc
@@ -0,0 +1,88 @@
+/*
+ * 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/accelerate.h"
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+Accelerate::ReturnCodes Accelerate::Process(
+ const int16_t* input,
+ size_t input_length,
+ AudioMultiVector* output,
+ int16_t* length_change_samples) {
+ // Input length must be (almost) 30 ms.
+ static const int k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate.
+ if (num_channels_ == 0 || static_cast<int>(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(input, input_length);
+ return kError;
+ }
+ return TimeStretch::Process(input, input_length, output,
+ length_change_samples);
+}
+
+void Accelerate::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;
+}
+
+Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
+ const int16_t* input, size_t input_length, size_t peak_index,
+ int16_t best_correlation, bool active_speech,
+ AudioMultiVector* output) const {
+ // Check for strong correlation or passive speech.
+ if ((best_correlation > kCorrelationThreshold) || !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;
+
+ assert(fs_mult_120 >= peak_index); // Should be handled in Process().
+ // Copy first part; 0 to 15 ms.
+ output->PushBackInterleaved(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(&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(
+ &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(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