Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/sync_buffer.cc b/webrtc/modules/audio_coding/neteq4/sync_buffer.cc
new file mode 100644
index 0000000..72fa16a
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/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/neteq4/sync_buffer.h"
+
+namespace webrtc {
+
+size_t SyncBuffer::FutureLength() const {
+ return Size() - next_index_;
+}
+
+void SyncBuffer::PushBack(const AudioMultiVector<int16_t>& append_this) {
+ size_t samples_added = append_this.Size();
+ AudioMultiVector<int16_t>::PushBack(append_this);
+ AudioMultiVector<int16_t>::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<int16_t>::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<int16_t>& 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<int16_t>::OverwriteAt(insert_this, length, position);
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector<int16_t>& 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