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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h

Issue 1802993002: Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_7
Patch Set: Reanimate CreateAudioReceiver() with 5 params, to not break downstream code. Created 4 years, 8 months ago
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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,
(...skipping 54 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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