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

webrtc/modules/audio_coding/neteq/sync_buffer.cc

@@ -0,0 +1,107 @@
+/*
+ *  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 <assert.h>
+
+#include <algorithm>  // Access to min.
+
+#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+
+namespace webrtc {
+
+size_t SyncBuffer::FutureLength() const {
+  return Size() - next_index_;
+}
+
+void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
+  size_t samples_added = append_this.Size();
+  AudioMultiVector::PushBack(append_this);
+  AudioMultiVector::PopFront(samples_added);
+  if (samples_added <= next_index_) {
+    next_index_ -= samples_added;
+  } else {
+    // This means that we are pushing out future data that was never used.
+//    assert(false);
+    // TODO(hlundin): This assert must be disabled to support 60 ms frames.
+    // This should not happen even for 60 ms frames, but it does. Investigate
+    // why.
+    next_index_ = 0;
+  }
+  dtmf_index_ -= std::min(dtmf_index_, samples_added);
+}
+
+void SyncBuffer::PushFrontZeros(size_t length) {
+  InsertZerosAtIndex(length, 0);
+}
+
+void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
+  position = std::min(position, Size());
+  length = std::min(length, Size() - position);
+  AudioMultiVector::PopBack(length);
+  for (size_t channel = 0; channel < Channels(); ++channel) {
+    channels_[channel]->InsertZerosAt(length, position);
+  }
+  if (next_index_ >= position) {
+    // We are moving the |next_index_| sample.
+    set_next_index(next_index_ + length);  // Overflow handled by subfunction.
+  }
+  if (dtmf_index_ > 0 && dtmf_index_ >= position) {
+    // We are moving the |dtmf_index_| sample.
+    set_dtmf_index(dtmf_index_ + length);  // Overflow handled by subfunction.
+  }
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t length,
+                                size_t position) {
+  position = std::min(position, Size());  // Cap |position| in the valid range.
+  length = std::min(length, Size() - position);
+  AudioMultiVector::OverwriteAt(insert_this, length, position);
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t position) {
+  ReplaceAtIndex(insert_this, insert_this.Size(), position);
+}
+
+size_t SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
+                                           int16_t* output) {
+  if (!output) {
+    assert(false);
+    return 0;
+  }
+  size_t samples_to_read = std::min(FutureLength(), requested_len);
+  ReadInterleavedFromIndex(next_index_, samples_to_read, output);
+  next_index_ += samples_to_read;
+  return samples_to_read;
+}
+
+void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
+  end_timestamp_ += increment;
+}
+
+void SyncBuffer::Flush() {
+  Zeros(Size());
+  next_index_ = Size();
+  end_timestamp_ = 0;
+  dtmf_index_ = 0;
+}
+
+void SyncBuffer::set_next_index(size_t value) {
+  // Cannot set |next_index_| larger than the size of the buffer.
+  next_index_ = std::min(value, Size());
+}
+
+void SyncBuffer::set_dtmf_index(size_t value) {
+  // Cannot set |dtmf_index_| larger than the size of the buffer.
+  dtmf_index_ = std::min(value, Size());
+}
+
+}  // namespace webrtc