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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
59 lines
1.7 KiB
C++
59 lines
1.7 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 "test/mock_audio_encoder.h"
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namespace webrtc {
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MockAudioEncoder::MockAudioEncoder() = default;
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MockAudioEncoder::~MockAudioEncoder() {
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Die();
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}
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MockAudioEncoder::FakeEncoding::FakeEncoding(
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const AudioEncoder::EncodedInfo& info)
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: info_(info) {}
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MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) {
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info_.encoded_bytes = encoded_bytes;
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}
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AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
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uint32_t timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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encoded->SetSize(encoded->size() + info_.encoded_bytes);
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return info_;
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}
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MockAudioEncoder::CopyEncoding::~CopyEncoding() = default;
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MockAudioEncoder::CopyEncoding::CopyEncoding(
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AudioEncoder::EncodedInfo info,
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rtc::ArrayView<const uint8_t> payload)
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: info_(info), payload_(payload) {}
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MockAudioEncoder::CopyEncoding::CopyEncoding(
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rtc::ArrayView<const uint8_t> payload)
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: payload_(payload) {
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info_.encoded_bytes = payload_.size();
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}
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AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
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uint32_t timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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RTC_CHECK(encoded);
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RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
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encoded->AppendData(payload_.data(), info_.encoded_bytes);
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return info_;
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}
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} // namespace webrtc
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