Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(194)

Side by Side Diff: webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase, again Created 5 years, 4 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
OLDNEW
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_INTERFACE_RTP_RTCP_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
12 #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
13 13
14 #include <set> 14 #include <set>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/modules/interface/module.h" 17 #include "webrtc/modules/interface/module.h"
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
19 19
20 namespace webrtc { 20 namespace webrtc {
21 // Forward declarations. 21 // Forward declarations.
22 class PacedSender; 22 class PacedSender;
23 class PacketRouter;
23 class ReceiveStatistics; 24 class ReceiveStatistics;
24 class RemoteBitrateEstimator; 25 class RemoteBitrateEstimator;
25 class RtpReceiver; 26 class RtpReceiver;
26 class Transport; 27 class Transport;
27 28
28 class RtpRtcp : public Module { 29 class RtpRtcp : public Module {
29 public: 30 public:
30 struct Configuration { 31 struct Configuration {
31 Configuration(); 32 Configuration();
32 33
(...skipping 21 matching lines...) Expand all
54 * bursts to minimize packet loss. 55 * bursts to minimize packet loss.
55 */ 56 */
56 int32_t id; 57 int32_t id;
57 bool audio; 58 bool audio;
58 bool receiver_only; 59 bool receiver_only;
59 Clock* clock; 60 Clock* clock;
60 ReceiveStatistics* receive_statistics; 61 ReceiveStatistics* receive_statistics;
61 Transport* outgoing_transport; 62 Transport* outgoing_transport;
62 RtcpIntraFrameObserver* intra_frame_callback; 63 RtcpIntraFrameObserver* intra_frame_callback;
63 RtcpBandwidthObserver* bandwidth_callback; 64 RtcpBandwidthObserver* bandwidth_callback;
65 SendTimeObserver* send_time_callback;
64 RtcpRttStats* rtt_stats; 66 RtcpRttStats* rtt_stats;
65 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; 67 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
66 RtpAudioFeedback* audio_messages; 68 RtpAudioFeedback* audio_messages;
67 RemoteBitrateEstimator* remote_bitrate_estimator; 69 RemoteBitrateEstimator* remote_bitrate_estimator;
68 PacedSender* paced_sender; 70 PacedSender* paced_sender;
71 PacketRouter* packet_router;
69 BitrateStatisticsObserver* send_bitrate_observer; 72 BitrateStatisticsObserver* send_bitrate_observer;
70 FrameCountObserver* send_frame_count_observer; 73 FrameCountObserver* send_frame_count_observer;
71 SendSideDelayObserver* send_side_delay_observer; 74 SendSideDelayObserver* send_side_delay_observer;
72 }; 75 };
73 76
74 /* 77 /*
75 * Create a RTP/RTCP module object using the system clock. 78 * Create a RTP/RTCP module object using the system clock.
76 * 79 *
77 * configuration - Configuration of the RTP/RTCP module. 80 * configuration - Configuration of the RTP/RTCP module.
78 */ 81 */
(...skipping 548 matching lines...) Expand 10 before | Expand all | Expand 10 after
627 630
628 /* 631 /*
629 * send a request for a keyframe 632 * send a request for a keyframe
630 * 633 *
631 * return -1 on failure else 0 634 * return -1 on failure else 0
632 */ 635 */
633 virtual int32_t RequestKeyFrame() = 0; 636 virtual int32_t RequestKeyFrame() = 0;
634 }; 637 };
635 } // namespace webrtc 638 } // namespace webrtc
636 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ 639 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698