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/*
* 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_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/modules/video_coding/main/test/video_source.h"
#include "webrtc/typedefs.h"
#include <stdio.h>
#include <string>
class RtpDataCallback : public webrtc::NullRtpData {
public:
RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
virtual ~RtpDataCallback() {}
virtual int32_t OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) OVERRIDE {
return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
}
private:
webrtc::VideoCodingModule* vcm_;
};
int RtpPlay(const CmdArgs& args);
int RtpPlayMT(const CmdArgs& args);
int ReceiverTimingTests(CmdArgs& args);
int JitterBufferTest(CmdArgs& args);
int DecodeFromStorageTest(const CmdArgs& args);
// Thread functions:
bool ProcessingThread(void* obj);
bool RtpReaderThread(void* obj);
bool DecodeThread(void* obj);
bool NackThread(void* obj);
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
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