4fb3d2bcca
They can all benefit from moving, since they contain std::string and std::vector. We intended to add these in https://codereview.webrtc.org/1896953004/, but got compiler errors we couldn't make sense of, so we skipped them. It turns out that what the compiler was complaining about was that when we said we'd have a user-defined move constructor, it stopped generating a copy assignment operator for us. This CL solves the problem by outfitting the types with defaulted copy and move assignment operators too. Review URL: https://codereview.webrtc.org/1899173002 Cr-Commit-Position: refs/heads/master@{#12469}
69 lines
2.0 KiB
C++
69 lines
2.0 KiB
C++
/*
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* Copyright (c) 2014 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/modules/audio_coding/codecs/audio_encoder.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/trace_event.h"
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namespace webrtc {
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AudioEncoder::EncodedInfo::EncodedInfo() = default;
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AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
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AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
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AudioEncoder::EncodedInfo::~EncodedInfo() = default;
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AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
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const EncodedInfo&) = default;
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AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
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default;
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int AudioEncoder::RtpTimestampRateHz() const {
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return SampleRateHz();
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}
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AudioEncoder::EncodedInfo AudioEncoder::Encode(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
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RTC_CHECK_EQ(audio.size(),
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static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
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const size_t old_size = encoded->size();
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EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
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RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
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return info;
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}
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bool AudioEncoder::SetFec(bool enable) {
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return !enable;
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}
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bool AudioEncoder::SetDtx(bool enable) {
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return !enable;
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}
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bool AudioEncoder::SetApplication(Application application) {
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return false;
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}
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void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
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void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
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void AudioEncoder::SetTargetBitrate(int target_bps) {}
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size_t AudioEncoder::MaxEncodedBytes() const {
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RTC_CHECK(false);
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return 0;
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}
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} // namespace webrtc
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