2557b86e76
The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
98 lines
3.0 KiB
C++
98 lines
3.0 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_
|