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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
13 | 13 |
14 #include "webrtc/base/scoped_ptr.h" | 14 #include "webrtc/base/scoped_ptr.h" |
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 17 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
19 #include "webrtc/typedefs.h" | 19 #include "webrtc/typedefs.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
22 | 22 |
23 class TelephoneEventHandler; | 23 class TelephoneEventHandler; |
24 | 24 |
25 // This strategy deals with media-specific RTP packet processing. | 25 // This strategy deals with media-specific RTP packet processing. |
26 // This class is not thread-safe and must be protected by its caller. | 26 // This class is not thread-safe and must be protected by its caller. |
27 class RTPReceiverStrategy { | 27 class RTPReceiverStrategy { |
28 public: | 28 public: |
29 static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); | 29 static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback); |
30 static RTPReceiverStrategy* CreateAudioStrategy( | 30 static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback); |
31 RtpData* data_callback, | |
32 RtpAudioFeedback* incoming_messages_callback); | |
33 | 31 |
34 virtual ~RTPReceiverStrategy() {} | 32 virtual ~RTPReceiverStrategy() {} |
35 | 33 |
36 // Parses the RTP packet and calls the data callback with the payload data. | 34 // Parses the RTP packet and calls the data callback with the payload data. |
37 // Implementations are encouraged to use the provided packet buffer and RTP | 35 // Implementations are encouraged to use the provided packet buffer and RTP |
38 // header as arguments to the callback; implementations are also allowed to | 36 // header as arguments to the callback; implementations are also allowed to |
39 // make changes in the data as necessary. The specific_payload argument | 37 // make changes in the data as necessary. The specific_payload argument |
40 // provides audio or video-specific data. The is_first_packet argument is true | 38 // provides audio or video-specific data. The is_first_packet argument is true |
41 // if this packet is either the first packet ever or the first in its frame. | 39 // if this packet is either the first packet ever or the first in its frame. |
42 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, | 40 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, |
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97 // packet. | 95 // packet. |
98 explicit RTPReceiverStrategy(RtpData* data_callback); | 96 explicit RTPReceiverStrategy(RtpData* data_callback); |
99 | 97 |
100 rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; | 98 rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_; |
101 PayloadUnion last_payload_; | 99 PayloadUnion last_payload_; |
102 RtpData* data_callback_; | 100 RtpData* data_callback_; |
103 }; | 101 }; |
104 } // namespace webrtc | 102 } // namespace webrtc |
105 | 103 |
106 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ | 104 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ |
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