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/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
new file mode 100644
index 0000000..1acf951
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -0,0 +1,187 @@
+/*
+ * 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_EXPAND_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+
+#include <assert.h>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class BackgroundNoise;
+class RandomVector;
+class SyncBuffer;
+
+// This class handles extrapolation of audio data from the sync_buffer to
+// produce packet-loss concealment.
+// TODO(hlundin): Refactor this class to divide the long methods into shorter
+// ones.
+class Expand {
+ public:
+ Expand(BackgroundNoise* background_noise,
+ SyncBuffer* sync_buffer,
+ RandomVector* random_vector,
+ int fs,
+ size_t num_channels)
+ : random_vector_(random_vector),
+ sync_buffer_(sync_buffer),
+ first_expand_(true),
+ fs_hz_(fs),
+ num_channels_(num_channels),
+ consecutive_expands_(0),
+ background_noise_(background_noise),
+ overlap_length_(5 * fs / 8000),
+ lag_index_direction_(0),
+ current_lag_index_(0),
+ stop_muting_(false),
+ channel_parameters_(new ChannelParameters[num_channels_]) {
+ assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
+ assert(fs <= kMaxSampleRate); // Should not be possible.
+ assert(num_channels_ > 0);
+ memset(expand_lags_, 0, sizeof(expand_lags_));
+ Reset();
+ }
+
+ virtual ~Expand() {}
+
+ // Resets the object.
+ virtual void Reset();
+
+ // The main method to produce concealment data. The data is appended to the
+ // end of |output|.
+ virtual int Process(AudioMultiVector* output);
+
+ // Prepare the object to do extra expansion during normal operation following
+ // a period of expands.
+ virtual void SetParametersForNormalAfterExpand();
+
+ // Prepare the object to do extra expansion during merge operation following
+ // a period of expands.
+ virtual void SetParametersForMergeAfterExpand();
+
+ // Sets the mute factor for |channel| to |value|.
+ void SetMuteFactor(int16_t value, size_t channel) {
+ assert(channel < num_channels_);
+ channel_parameters_[channel].mute_factor = value;
+ }
+
+ // Returns the mute factor for |channel|.
+ int16_t MuteFactor(size_t channel) {
+ assert(channel < num_channels_);
+ return channel_parameters_[channel].mute_factor;
+ }
+
+ // Accessors and mutators.
+ virtual size_t overlap_length() const { return overlap_length_; }
+ int16_t max_lag() const { return max_lag_; }
+
+ protected:
+ static const int kMaxConsecutiveExpands = 200;
+ void GenerateRandomVector(int seed_increment,
+ size_t length,
+ int16_t* random_vector);
+
+ void GenerateBackgroundNoise(int16_t* random_vector,
+ size_t channel,
+ int16_t mute_slope,
+ bool too_many_expands,
+ size_t num_noise_samples,
+ int16_t* buffer);
+
+ // Initializes member variables at the beginning of an expand period.
+ void InitializeForAnExpandPeriod();
+
+ bool TooManyExpands();
+
+ // Analyzes the signal history in |sync_buffer_|, and set up all parameters
+ // necessary to produce concealment data.
+ void AnalyzeSignal(int16_t* random_vector);
+
+ RandomVector* random_vector_;
+ SyncBuffer* sync_buffer_;
+ bool first_expand_;
+ const int fs_hz_;
+ const size_t num_channels_;
+ int consecutive_expands_;
+
+ private:
+ static const int kUnvoicedLpcOrder = 6;
+ static const int kNumCorrelationCandidates = 3;
+ static const int kDistortionLength = 20;
+ static const int kLpcAnalysisLength = 160;
+ static const int kMaxSampleRate = 48000;
+ static const int kNumLags = 3;
+
+ struct ChannelParameters {
+ // Constructor.
+ ChannelParameters()
+ : mute_factor(16384),
+ ar_gain(0),
+ ar_gain_scale(0),
+ voice_mix_factor(0),
+ current_voice_mix_factor(0),
+ onset(false),
+ mute_slope(0) {
+ memset(ar_filter, 0, sizeof(ar_filter));
+ memset(ar_filter_state, 0, sizeof(ar_filter_state));
+ }
+ int16_t mute_factor;
+ int16_t ar_filter[kUnvoicedLpcOrder + 1];
+ int16_t ar_filter_state[kUnvoicedLpcOrder];
+ int16_t ar_gain;
+ int16_t ar_gain_scale;
+ int16_t voice_mix_factor; /* Q14 */
+ int16_t current_voice_mix_factor; /* Q14 */
+ AudioVector expand_vector0;
+ AudioVector expand_vector1;
+ bool onset;
+ int16_t mute_slope; /* Q20 */
+ };
+
+ // Calculate the auto-correlation of |input|, with length |input_length|
+ // samples. The correlation is calculated from a downsampled version of
+ // |input|, and is written to |output|. The scale factor is written to
+ // |output_scale|. Returns the length of the correlation vector.
+ int16_t Correlation(const int16_t* input, size_t input_length,
+ int16_t* output, int16_t* output_scale) const;
+
+ void UpdateLagIndex();
+
+ BackgroundNoise* background_noise_;
+ const size_t overlap_length_;
+ int16_t max_lag_;
+ size_t expand_lags_[kNumLags];
+ int lag_index_direction_;
+ int current_lag_index_;
+ bool stop_muting_;
+ scoped_ptr<ChannelParameters[]> channel_parameters_;
+
+ DISALLOW_COPY_AND_ASSIGN(Expand);
+};
+
+struct ExpandFactory {
+ ExpandFactory() {}
+ virtual ~ExpandFactory() {}
+
+ virtual Expand* Create(BackgroundNoise* background_noise,
+ SyncBuffer* sync_buffer,
+ RandomVector* random_vector,
+ int fs,
+ size_t num_channels) const;
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_