They are invoked with the maximum size of the data to be added, and a
callable that generates that data, like this:
buffer.AppendData(10, [] (rtc::ArrayView<uint8_t> av) {
for (uint8_t i = 0; i != 5; ++i)
av[i] = i;
return 5;
});
The callable returns the number of bytes actually written, and the
final Buffer size will be adjusted accordingly. SetData and AppendData
both return the number of bytes added (i.e. the return value of the
callable).
These versions will be useful when converting AudioEncoder::Encode to use Buffer rather than raw pointers.
Also added a few tests for the new functionality.
Review URL: https://codereview.webrtc.org/1717273002
Cr-Commit-Position: refs/heads/master@{#11733}
48 lines
1.2 KiB
C++
48 lines
1.2 KiB
C++
/*
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* Copyright 2015 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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#include "webrtc/base/buffer.h"
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#include <algorithm>
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#include <cassert>
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#include <utility>
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namespace rtc {
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Buffer::Buffer() : size_(0), capacity_(0), data_(nullptr) {
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assert(IsConsistent());
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}
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Buffer::Buffer(const Buffer& buf) : Buffer(buf.data(), buf.size()) {
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}
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Buffer::Buffer(Buffer&& buf)
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: size_(buf.size()),
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capacity_(buf.capacity()),
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data_(std::move(buf.data_)) {
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assert(IsConsistent());
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buf.OnMovedFrom();
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}
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Buffer::Buffer(size_t size) : Buffer(size, size) {
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}
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Buffer::Buffer(size_t size, size_t capacity)
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: size_(size),
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capacity_(std::max(size, capacity)),
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data_(new uint8_t[capacity_]) {
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assert(IsConsistent());
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}
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// Note: The destructor works even if the buffer has been moved from.
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Buffer::~Buffer() = default;
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}; // namespace rtc
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