Files
platform-external-webrtc/webrtc/modules/video_coding/test/rtp_player.h
philipel 5908c71128 Lint fix for webrtc/modules/video_coding PART 3!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1540243002

Cr-Commit-Position: refs/heads/master@{#11105}
2015-12-21 16:23:29 +00:00

101 lines
3.1 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_VIDEO_CODING_TEST_RTP_PLAYER_H_
#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
#include <string>
#include <vector>
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
namespace webrtc {
class Clock;
namespace rtpplayer {
class PayloadCodecTuple {
public:
PayloadCodecTuple(uint8_t payload_type,
const std::string& codec_name,
VideoCodecType codec_type)
: name_(codec_name),
payload_type_(payload_type),
codec_type_(codec_type) {}
const std::string& name() const { return name_; }
uint8_t payload_type() const { return payload_type_; }
VideoCodecType codec_type() const { return codec_type_; }
private:
std::string name_;
uint8_t payload_type_;
VideoCodecType codec_type_;
};
typedef std::vector<PayloadCodecTuple> PayloadTypes;
typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator;
// Implemented by RtpPlayer and given to client as a means to retrieve
// information about a specific RTP stream.
class RtpStreamInterface {
public:
virtual ~RtpStreamInterface() {}
// Ask for missing packets to be resent.
virtual void ResendPackets(const uint16_t* sequence_numbers,
uint16_t length) = 0;
virtual uint32_t ssrc() const = 0;
virtual const PayloadTypes& payload_types() const = 0;
};
// Implemented by a sink. Wraps RtpData because its d-tor is protected.
class PayloadSinkInterface : public RtpData {
public:
virtual ~PayloadSinkInterface() {}
};
// Implemented to provide a sink for RTP data, such as hooking up a VCM to
// the incoming RTP stream.
class PayloadSinkFactoryInterface {
public:
virtual ~PayloadSinkFactoryInterface() {}
// Return NULL if failed to create sink. 'stream' is guaranteed to be
// around for as long as the RtpData. The returned object is owned by
// the caller (RtpPlayer).
virtual PayloadSinkInterface* Create(RtpStreamInterface* stream) = 0;
};
// The client's view of an RtpPlayer.
class RtpPlayerInterface {
public:
virtual ~RtpPlayerInterface() {}
virtual int NextPacket(int64_t timeNow) = 0;
virtual uint32_t TimeUntilNextPacket() const = 0;
virtual void Print() const = 0;
};
RtpPlayerInterface* Create(const std::string& inputFilename,
PayloadSinkFactoryInterface* payloadSinkFactory,
Clock* clock,
const PayloadTypes& payload_types,
float lossRate,
int64_t rttMs,
bool reordering);
} // namespace rtpplayer
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_