a22a9bd9ca
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch. BUG= TEST=vie & voe_auto_test full runs Review URL: https://webrtc-codereview.appspot.com/1014006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
33 lines
1.0 KiB
C++
33 lines
1.0 KiB
C++
/*
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* Copyright (c) 2012 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/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include <cstdlib>
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namespace webrtc {
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RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
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: data_callback_(data_callback) {
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memset(&last_payload_, 0, sizeof(last_payload_));
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}
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void RTPReceiverStrategy::GetLastMediaSpecificPayload(
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ModuleRTPUtility::PayloadUnion* payload) const {
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memcpy(payload, &last_payload_, sizeof(*payload));
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}
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void RTPReceiverStrategy::SetLastMediaSpecificPayload(
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const ModuleRTPUtility::PayloadUnion& payload) {
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memcpy(&last_payload_, &payload, sizeof(last_payload_));
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}
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} // namespace webrtc
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