| /* | 
 |  *  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/modules/include/module_common_types.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) {} | 
 |  | 
 |   // 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) override; | 
 |  | 
 |   // 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. The AudioFrame |output| is first reset, and the |data_|, | 
 |   // |num_channels_|, and |samples_per_channel_| fields are updated. | 
 |   void GetNextAudioInterleaved(size_t requested_len, AudioFrame* 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. | 
 |  | 
 |   RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer); | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 | #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_ |