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phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +00001/*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
12#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +000013
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020014#include "modules/rtp_rtcp/include/rtp_rtcp.h"
15#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
16#include "modules/rtp_rtcp/source/rtp_utility.h"
17#include "rtc_base/criticalsection.h"
phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +000018
19namespace webrtc {
20
magjed56124bd2016-11-24 09:34:46 -080021struct CodecInst;
22
phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +000023// This strategy deals with media-specific RTP packet processing.
24// This class is not thread-safe and must be protected by its caller.
25class RTPReceiverStrategy {
26 public:
andresp@webrtc.orgdc80bae2014-04-08 11:06:12 +000027 static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
solenberg1d031392016-03-30 02:42:32 -070028 static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
wu@webrtc.org822fbd82013-08-15 23:38:54 +000029
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010030 virtual ~RTPReceiverStrategy();
stefan@webrtc.org66b2e5c2013-07-05 14:30:48 +000031
tnakamura@webrtc.orgaa4d96a2013-07-16 19:25:04 +000032 // Parses the RTP packet and calls the data callback with the payload data.
33 // Implementations are encouraged to use the provided packet buffer and RTP
34 // header as arguments to the callback; implementations are also allowed to
35 // make changes in the data as necessary. The specific_payload argument
Niels Möllerbbf389c2017-09-26 14:05:05 +020036 // provides audio or video-specific data.
wu@webrtc.org822fbd82013-08-15 23:38:54 +000037 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
38 const PayloadUnion& specific_payload,
stefan@webrtc.org7bb8f022013-09-06 13:40:11 +000039 const uint8_t* payload,
pkasting@chromium.org4591fbd2014-11-20 22:28:14 +000040 size_t payload_length,
Niels Möllerbbf389c2017-09-26 14:05:05 +020041 int64_t timestamp_ms) = 0;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000042
tnakamura@webrtc.orgaa4d96a2013-07-16 19:25:04 +000043 protected:
wu@webrtc.org822fbd82013-08-15 23:38:54 +000044 // The data callback is where we should send received payload data.
45 // See ParseRtpPacket. This class does not claim ownership of the callback.
46 // Implementations must NOT hold any critical sections while calling the
47 // callback.
48 //
49 // Note: Implementations may call the callback for other reasons than calls
50 // to ParseRtpPacket, for instance if the implementation somehow recovers a
51 // packet.
danilchap6db6cdc2015-12-15 02:54:47 -080052 explicit RTPReceiverStrategy(RtpData* data_callback);
wu@webrtc.org822fbd82013-08-15 23:38:54 +000053
danilchap7c9426c2016-04-14 03:05:31 -070054 rtc::CriticalSection crit_sect_;
phoglund@webrtc.orga22a9bd2013-01-14 10:01:55 +000055 RtpData* data_callback_;
phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +000056};
phoglund@webrtc.org07bf43c2012-12-18 15:40:53 +000057} // namespace webrtc
58
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020059#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_