Files
platform-external-webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00

46 lines
1.4 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include <stdlib.h>
namespace webrtc {
RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
: data_callback_(data_callback) {
memset(&last_payload_, 0, sizeof(last_payload_));
}
void RTPReceiverStrategy::GetLastMediaSpecificPayload(
PayloadUnion* payload) const {
rtc::CritScope cs(&crit_sect_);
memcpy(payload, &last_payload_, sizeof(*payload));
}
void RTPReceiverStrategy::SetLastMediaSpecificPayload(
const PayloadUnion& payload) {
rtc::CritScope cs(&crit_sect_);
memcpy(&last_payload_, &payload, sizeof(last_payload_));
}
void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
PayloadUnion* specific_payload,
bool* should_discard_changes) {
// Default: Keep changes.
*should_discard_changes = false;
}
int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
return -1;
}
} // namespace webrtc