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604fd2f1ab
Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771}
110 lines
4.6 KiB
C++
110 lines
4.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "api/audio/audio_frame.h"
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#include "modules/audio_coding/neteq/audio_multi_vector.h"
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#include "modules/audio_coding/neteq/audio_vector.h"
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#include "rtc_base/buffer.h"
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namespace webrtc {
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class SyncBuffer : public AudioMultiVector {
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public:
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SyncBuffer(size_t channels, size_t length)
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: AudioMultiVector(channels, length),
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next_index_(length),
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end_timestamp_(0),
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dtmf_index_(0) {}
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SyncBuffer(const SyncBuffer&) = delete;
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SyncBuffer& operator=(const SyncBuffer&) = delete;
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// Returns the number of samples yet to play out from the buffer.
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size_t FutureLength() const;
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// Adds the contents of `append_this` to the back of the SyncBuffer. Removes
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// the same number of samples from the beginning of the SyncBuffer, to
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// maintain a constant buffer size. The `next_index_` is updated to reflect
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// the move of the beginning of "future" data.
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void PushBack(const AudioMultiVector& append_this) override;
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// Like PushBack, but reads the samples channel-interleaved from the input.
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void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
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// Adds `length` zeros to the beginning of each channel. Removes
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// the same number of samples from the end of the SyncBuffer, to
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// maintain a constant buffer size. The `next_index_` is updated to reflect
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// the move of the beginning of "future" data.
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// Note that this operation may delete future samples that are waiting to
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// be played.
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void PushFrontZeros(size_t length);
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// Inserts `length` zeros into each channel at index `position`. The size of
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// the SyncBuffer is kept constant, which means that the last `length`
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// elements in each channel will be purged.
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virtual void InsertZerosAtIndex(size_t length, size_t position);
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// Overwrites each channel in this SyncBuffer with values taken from
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// `insert_this`. The values are taken from the beginning of `insert_this` and
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// are inserted starting at `position`. `length` values are written into each
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// channel. The size of the SyncBuffer is kept constant. That is, if `length`
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// and `position` are selected such that the new data would extend beyond the
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// end of the current SyncBuffer, the buffer is not extended.
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// The `next_index_` is not updated.
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virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t length,
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size_t position);
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// Same as the above method, but where all of `insert_this` is written (with
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// the same constraints as above, that the SyncBuffer is not extended).
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virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t position);
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// Reads `requested_len` samples from each channel and writes them interleaved
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// into `output`. The `next_index_` is updated to point to the sample to read
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// next time. The AudioFrame `output` is first reset, and the `data_`,
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// `num_channels_`, and `samples_per_channel_` fields are updated.
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void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
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// Adds `increment` to `end_timestamp_`.
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void IncreaseEndTimestamp(uint32_t increment);
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// Flushes the buffer. The buffer will contain only zeros after the flush, and
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// `next_index_` will point to the end, like when the buffer was first
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// created.
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void Flush();
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const AudioVector& Channel(size_t n) const { return *channels_[n]; }
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AudioVector& Channel(size_t n) { return *channels_[n]; }
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// Accessors and mutators.
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size_t next_index() const { return next_index_; }
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void set_next_index(size_t value);
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uint32_t end_timestamp() const { return end_timestamp_; }
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void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
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size_t dtmf_index() const { return dtmf_index_; }
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void set_dtmf_index(size_t value);
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private:
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size_t next_index_;
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uint32_t end_timestamp_; // The timestamp of the last sample in the buffer.
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size_t dtmf_index_; // Index to the first non-DTMF sample in the buffer.
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
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