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/sync_buffer.h b/webrtc/modules/audio_coding/neteq/sync_buffer.h
new file mode 100644
index 0000000..59bd4d8
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -0,0 +1,101 @@
+/*
+ * 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_SYNC_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class SyncBuffer : public AudioMultiVector {
+ public:
+ SyncBuffer(size_t channels, size_t length)
+ : AudioMultiVector(channels, length),
+ next_index_(length),
+ end_timestamp_(0),
+ dtmf_index_(0) {}
+
+ virtual ~SyncBuffer() {}
+
+ // Returns the number of samples yet to play out form the buffer.
+ size_t FutureLength() const;
+
+ // Adds the contents of |append_this| to the back of the SyncBuffer. Removes
+ // the same number of samples from the beginning of the SyncBuffer, to
+ // maintain a constant buffer size. The |next_index_| is updated to reflect
+ // the move of the beginning of "future" data.
+ void PushBack(const AudioMultiVector& append_this);
+
+ // Adds |length| zeros to the beginning of each channel. Removes
+ // the same number of samples from the end of the SyncBuffer, to
+ // maintain a constant buffer size. The |next_index_| is updated to reflect
+ // the move of the beginning of "future" data.
+ // Note that this operation may delete future samples that are waiting to
+ // be played.
+ void PushFrontZeros(size_t length);
+
+ // Inserts |length| zeros into each channel at index |position|. The size of
+ // the SyncBuffer is kept constant, which means that the last |length|
+ // elements in each channel will be purged.
+ virtual void InsertZerosAtIndex(size_t length, size_t position);
+
+ // Overwrites each channel in this SyncBuffer with values taken from
+ // |insert_this|. The values are taken from the beginning of |insert_this| and
+ // are inserted starting at |position|. |length| values are written into each
+ // channel. The size of the SyncBuffer is kept constant. That is, if |length|
+ // and |position| are selected such that the new data would extend beyond the
+ // end of the current SyncBuffer, the buffer is not extended.
+ // The |next_index_| is not updated.
+ virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+ size_t length,
+ size_t position);
+
+ // Same as the above method, but where all of |insert_this| is written (with
+ // the same constraints as above, that the SyncBuffer is not extended).
+ virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+ size_t position);
+
+ // Reads |requested_len| samples from each channel and writes them interleaved
+ // into |output|. The |next_index_| is updated to point to the sample to read
+ // next time.
+ size_t GetNextAudioInterleaved(size_t requested_len, int16_t* output);
+
+ // Adds |increment| to |end_timestamp_|.
+ void IncreaseEndTimestamp(uint32_t increment);
+
+ // Flushes the buffer. The buffer will contain only zeros after the flush, and
+ // |next_index_| will point to the end, like when the buffer was first
+ // created.
+ void Flush();
+
+ const AudioVector& Channel(size_t n) const { return *channels_[n]; }
+ AudioVector& Channel(size_t n) { return *channels_[n]; }
+
+ // Accessors and mutators.
+ size_t next_index() const { return next_index_; }
+ void set_next_index(size_t value);
+ uint32_t end_timestamp() const { return end_timestamp_; }
+ void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
+ size_t dtmf_index() const { return dtmf_index_; }
+ void set_dtmf_index(size_t value);
+
+ private:
+ size_t next_index_;
+ uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
+ size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
+
+ DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_