terelius 838cdb3db6 Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ )
Reason for revert:
Broke internal project

Original issue's description:
> Fix chromium-style warnings.
>
> Separate the null implementation from rtp_rtcp_defines.h, and follow chromium style guide for virtual functions.
>
> BUG=webrtc:163
>
> Committed: https://crrev.com/509eadd554de6bf938da08071c5d2c2541703134
> Cr-Commit-Position: refs/heads/master@{#14738}

TBR=danilchap@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:163

Review-Url: https://codereview.webrtc.org/2449523002
Cr-Commit-Position: refs/heads/master@{#14750}
2016-10-24 16:38:26 +00:00

73 lines
2.5 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.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
#define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/test/gtest.h"
#include "webrtc/transport.h"
namespace webrtc {
// This class sends all its packet straight to the provided RtpRtcp module.
// with optional packet loss.
class LoopBackTransport : public Transport {
public:
LoopBackTransport()
: count_(0),
packet_loss_(0),
rtp_payload_registry_(NULL),
rtp_receiver_(NULL),
rtp_rtcp_module_(NULL) {}
void SetSendModule(RtpRtcp* rtp_rtcp_module,
RTPPayloadRegistry* payload_registry,
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics);
void DropEveryNthPacket(int n);
bool SendRtp(const uint8_t* data,
size_t len,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* data, size_t len) override;
private:
int count_;
int packet_loss_;
ReceiveStatistics* receive_statistics_;
RTPPayloadRegistry* rtp_payload_registry_;
RtpReceiver* rtp_receiver_;
RtpRtcp* rtp_rtcp_module_;
};
class TestRtpReceiver : public NullRtpData {
public:
int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) override;
const uint8_t* payload_data() const { return payload_data_; }
size_t payload_size() const { return payload_size_; }
webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; }
private:
uint8_t payload_data_[1500];
size_t payload_size_;
webrtc::WebRtcRTPHeader rtp_header_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_