Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(89)

Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

Issue 3012253002: Reland of Delete Rtx-related methods from RTPPayloadRegistry. (Closed)
Patch Set: Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/modules/rtp_rtcp/BUILD.gn ('k') | webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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
(...skipping 35 matching lines...) Expand 10 before | Expand all | Expand 10 after
46 int8_t* payload_type) const; 46 int8_t* payload_type) const;
47 47
48 bool RtxEnabled() const; 48 bool RtxEnabled() const;
49 49
50 void SetRtxSsrc(uint32_t ssrc); 50 void SetRtxSsrc(uint32_t ssrc);
51 51
52 bool GetRtxSsrc(uint32_t* ssrc) const; 52 bool GetRtxSsrc(uint32_t* ssrc) const;
53 53
54 void SetRtxPayloadType(int payload_type, int associated_payload_type); 54 void SetRtxPayloadType(int payload_type, int associated_payload_type);
55 55
56 bool IsRtx(const RTPHeader& header) const;
57
58 bool RestoreOriginalPacket(uint8_t* restored_packet,
59 const uint8_t* packet,
60 size_t* packet_length,
61 uint32_t original_ssrc,
62 const RTPHeader& header);
63
64 bool IsRed(const RTPHeader& header) const; 56 bool IsRed(const RTPHeader& header) const;
65 57
66 // Returns true if the media of this RTP packet is encapsulated within an
67 // extra header, such as RTX or RED.
68 bool IsEncapsulated(const RTPHeader& header) const;
69
70 bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const; 58 bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
71 59
72 int GetPayloadTypeFrequency(uint8_t payload_type) const; 60 int GetPayloadTypeFrequency(uint8_t payload_type) const;
73 61
74 const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const; 62 const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
75 63
76 void ResetLastReceivedPayloadTypes() { 64 void ResetLastReceivedPayloadTypes() {
77 rtc::CritScope cs(&crit_sect_); 65 rtc::CritScope cs(&crit_sect_);
78 last_received_payload_type_ = -1; 66 last_received_payload_type_ = -1;
79 last_received_media_payload_type_ = -1; 67 last_received_media_payload_type_ = -1;
(...skipping 52 matching lines...) Expand 10 before | Expand all | Expand 10 after
132 // video, DCHECK that no instance is used for both audio and video. 120 // video, DCHECK that no instance is used for both audio and video.
133 #if RTC_DCHECK_IS_ON 121 #if RTC_DCHECK_IS_ON
134 bool used_for_audio_ GUARDED_BY(crit_sect_) = false; 122 bool used_for_audio_ GUARDED_BY(crit_sect_) = false;
135 bool used_for_video_ GUARDED_BY(crit_sect_) = false; 123 bool used_for_video_ GUARDED_BY(crit_sect_) = false;
136 #endif 124 #endif
137 }; 125 };
138 126
139 } // namespace webrtc 127 } // namespace webrtc
140 128
141 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ 129 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/BUILD.gn ('k') | webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698