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1c54605e77
This CL is a manual spin-off of [1], which tried to apply clang-tidy's performance-move-const-arg [1] to the WebRTC codebase. Since there were some wrong fixes to correct, this CL lands a few different fixes, like adding a constructor overload to take an rvalue reference or remove 'const' to make std::move effective. [1] - https://webrtc-review.googlesource.com/c/src/+/120350 [2] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html Bug: webrtc:10252 Change-Id: I42a777247fee2cb788efcd7c2035148330056b7a Reviewed-on: https://webrtc-review.googlesource.com/c/120928 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26553}
86 lines
3.1 KiB
C++
86 lines
3.1 KiB
C++
/*
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* Copyright (c) 2016 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 "api/audio_codecs/audio_format.h"
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#include <utility>
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#include "absl/strings/match.h"
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namespace webrtc {
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SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
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SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
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SdpAudioFormat::SdpAudioFormat(absl::string_view name,
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int clockrate_hz,
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size_t num_channels)
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: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
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SdpAudioFormat::SdpAudioFormat(absl::string_view name,
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int clockrate_hz,
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size_t num_channels,
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const Parameters& param)
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: name(name),
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clockrate_hz(clockrate_hz),
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num_channels(num_channels),
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parameters(param) {}
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SdpAudioFormat::SdpAudioFormat(absl::string_view name,
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int clockrate_hz,
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size_t num_channels,
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Parameters&& param)
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: name(name),
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clockrate_hz(clockrate_hz),
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num_channels(num_channels),
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parameters(std::move(param)) {}
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bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
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return absl::EqualsIgnoreCase(name, o.name) &&
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clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
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}
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SdpAudioFormat::~SdpAudioFormat() = default;
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SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
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SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
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bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
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return absl::EqualsIgnoreCase(a.name, b.name) &&
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a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
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a.parameters == b.parameters;
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}
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AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
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size_t num_channels,
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int bitrate_bps)
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: AudioCodecInfo(sample_rate_hz,
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num_channels,
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bitrate_bps,
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bitrate_bps,
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bitrate_bps) {}
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AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
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size_t num_channels,
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int default_bitrate_bps,
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int min_bitrate_bps,
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int max_bitrate_bps)
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: sample_rate_hz(sample_rate_hz),
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num_channels(num_channels),
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default_bitrate_bps(default_bitrate_bps),
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min_bitrate_bps(min_bitrate_bps),
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max_bitrate_bps(max_bitrate_bps) {
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RTC_DCHECK_GT(sample_rate_hz, 0);
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RTC_DCHECK_GT(num_channels, 0);
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RTC_DCHECK_GE(min_bitrate_bps, 0);
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RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
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RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
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}
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} // namespace webrtc
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