
This reverts commit 021512b76a872b04e803d61f46c740ed363d641b. Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly. Original change's description: > rtpsender interface: make pure virtual again > > after providing default implementations in Chromium tests > > BUG=None > > Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100 > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#37941} Bug: None Change-Id: I40f27c36819365fadae32032521f7e11184bee62 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484 Owners-Override: Andrey Logvin <landrey@google.com> Commit-Queue: Andrey Logvin <landrey@google.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Andrey Logvin <landrey@google.com> Cr-Commit-Position: refs/heads/main@{#37947}
37 lines
1.0 KiB
C++
37 lines
1.0 KiB
C++
/*
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* Copyright 2018 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/rtp_sender_interface.h"
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namespace webrtc {
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void RtpSenderInterface::SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {}
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rtc::scoped_refptr<FrameEncryptorInterface>
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RtpSenderInterface::GetFrameEncryptor() const {
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return nullptr;
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}
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std::vector<RtpEncodingParameters> RtpSenderInterface::init_send_encodings()
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const {
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return {};
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}
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rtc::scoped_refptr<DtlsTransportInterface> RtpSenderInterface::dtls_transport()
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const {
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return nullptr;
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}
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void RtpSenderInterface::SetEncoderToPacketizerFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {}
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} // namespace webrtc
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