| /* |
| * 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_ |