66b2e5c05a
rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
42 lines
1.3 KiB
C++
42 lines
1.3 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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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
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: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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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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PayloadUnion* payload) const {
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CriticalSectionScoped cs(crit_sect_.get());
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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 PayloadUnion& payload) {
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CriticalSectionScoped cs(crit_sect_.get());
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memcpy(&last_payload_, &payload, sizeof(last_payload_));
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}
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int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
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return 0;
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}
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} // namespace webrtc
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