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/time_stretch.h b/webrtc/modules/audio_coding/neteq/time_stretch.h
new file mode 100644
index 0000000..9396d8f
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/time_stretch.h
@@ -0,0 +1,111 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+
+#include <assert.h>
+#include <string.h> // memset, size_t
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class BackgroundNoise;
+
+// This is the base class for Accelerate and PreemptiveExpand. This class
+// cannot be instantiated, but must be used through either of the derived
+// classes.
+class TimeStretch {
+ public:
+ enum ReturnCodes {
+ kSuccess = 0,
+ kSuccessLowEnergy = 1,
+ kNoStretch = 2,
+ kError = -1
+ };
+
+ TimeStretch(int sample_rate_hz, size_t num_channels,
+ const BackgroundNoise& background_noise)
+ : sample_rate_hz_(sample_rate_hz),
+ fs_mult_(sample_rate_hz / 8000),
+ num_channels_(static_cast<int>(num_channels)),
+ master_channel_(0), // First channel is master.
+ background_noise_(background_noise),
+ max_input_value_(0) {
+ assert(sample_rate_hz_ == 8000 ||
+ sample_rate_hz_ == 16000 ||
+ sample_rate_hz_ == 32000 ||
+ sample_rate_hz_ == 48000);
+ assert(num_channels_ > 0);
+ assert(static_cast<int>(master_channel_) < num_channels_);
+ memset(auto_correlation_, 0, sizeof(auto_correlation_));
+ }
+
+ virtual ~TimeStretch() {}
+
+ // This method performs the processing common to both Accelerate and
+ // PreemptiveExpand.
+ ReturnCodes Process(const int16_t* input,
+ size_t input_len,
+ AudioMultiVector* output,
+ int16_t* length_change_samples);
+
+ protected:
+ // Sets the parameters |best_correlation| and |peak_index| to suitable
+ // values when the signal contains no active speech. This method must be
+ // implemented by the sub-classes.
+ virtual void SetParametersForPassiveSpeech(size_t input_length,
+ int16_t* best_correlation,
+ int* peak_index) const = 0;
+
+ // Checks the criteria for performing the time-stretching operation and,
+ // if possible, performs the time-stretching. This method must be implemented
+ // by the sub-classes.
+ virtual ReturnCodes CheckCriteriaAndStretch(
+ const int16_t* input, size_t input_length, size_t peak_index,
+ int16_t best_correlation, bool active_speech,
+ AudioMultiVector* output) const = 0;
+
+ static const int kCorrelationLen = 50;
+ static const int kLogCorrelationLen = 6; // >= log2(kCorrelationLen).
+ static const int kMinLag = 10;
+ static const int kMaxLag = 60;
+ static const int kDownsampledLen = kCorrelationLen + kMaxLag;
+ static const int kCorrelationThreshold = 14746; // 0.9 in Q14.
+
+ const int sample_rate_hz_;
+ const int fs_mult_; // Sample rate multiplier = sample_rate_hz_ / 8000.
+ const int num_channels_;
+ const size_t master_channel_;
+ const BackgroundNoise& background_noise_;
+ int16_t max_input_value_;
+ int16_t downsampled_input_[kDownsampledLen];
+ // Adding 1 to the size of |auto_correlation_| because of how it is used
+ // by the peak-detection algorithm.
+ int16_t auto_correlation_[kCorrelationLen + 1];
+
+ private:
+ // Calculates the auto-correlation of |downsampled_input_| and writes the
+ // result to |auto_correlation_|.
+ void AutoCorrelation();
+
+ // Performs a simple voice-activity detection based on the input parameters.
+ bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
+ int peak_index, int scaling) const;
+
+ DISALLOW_COPY_AND_ASSIGN(TimeStretch);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_