16825b1a82
Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
98 lines
3.0 KiB
C++
98 lines
3.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
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#include <string>
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#include <vector>
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
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namespace webrtc {
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class Clock;
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namespace rtpplayer {
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class PayloadCodecTuple {
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public:
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PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name,
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VideoCodecType codec_type)
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: name_(codec_name),
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payload_type_(payload_type),
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codec_type_(codec_type) {
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}
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const std::string& name() const { return name_; }
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uint8_t payload_type() const { return payload_type_; }
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VideoCodecType codec_type() const { return codec_type_; }
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private:
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std::string name_;
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uint8_t payload_type_;
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VideoCodecType codec_type_;
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};
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typedef std::vector<PayloadCodecTuple> PayloadTypes;
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typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator;
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// Implemented by RtpPlayer and given to client as a means to retrieve
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// information about a specific RTP stream.
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class RtpStreamInterface {
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public:
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virtual ~RtpStreamInterface() {}
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// Ask for missing packets to be resent.
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virtual void ResendPackets(const uint16_t* sequence_numbers,
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uint16_t length) = 0;
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virtual uint32_t ssrc() const = 0;
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virtual const PayloadTypes& payload_types() const = 0;
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};
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// Implemented by a sink. Wraps RtpData because its d-tor is protected.
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class PayloadSinkInterface : public RtpData {
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public:
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virtual ~PayloadSinkInterface() {}
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};
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// Implemented to provide a sink for RTP data, such as hooking up a VCM to
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// the incoming RTP stream.
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class PayloadSinkFactoryInterface {
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public:
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virtual ~PayloadSinkFactoryInterface() {}
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// Return NULL if failed to create sink. 'stream' is guaranteed to be
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// around for as long as the RtpData. The returned object is owned by
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// the caller (RtpPlayer).
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virtual PayloadSinkInterface* Create(RtpStreamInterface* stream) = 0;
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};
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// The client's view of an RtpPlayer.
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class RtpPlayerInterface {
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public:
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virtual ~RtpPlayerInterface() {}
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virtual int NextPacket(int64_t timeNow) = 0;
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virtual uint32_t TimeUntilNextPacket() const = 0;
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virtual void Print() const = 0;
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};
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RtpPlayerInterface* Create(const std::string& inputFilename,
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PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock,
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const PayloadTypes& payload_types, float lossRate, int64_t rttMs,
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bool reordering);
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} // namespace rtpplayer
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} // namespace webrtc
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#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
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