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

Issue 2491293002: Make FlexFEC packets paceable through RTPSender. (Closed)
Patch Set: Feedback response 3. Created 4 years, 1 month 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_RTCP_IMPL_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
13 13
14 #include <set> 14 #include <set>
15 #include <utility> 15 #include <utility>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/base/gtest_prod_util.h" 19 #include "webrtc/base/gtest_prod_util.h"
20 #include "webrtc/base/optional.h"
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
22 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h" 23 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
24 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" 25 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
25 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" 26 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
26 27
27 namespace webrtc { 28 namespace webrtc {
28 29
29 class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp { 30 class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
(...skipping 60 matching lines...) Expand 10 before | Expand all | Expand 10 after
90 RTCPSender::FeedbackState GetFeedbackState(); 91 RTCPSender::FeedbackState GetFeedbackState();
91 92
92 void SetRtxSendStatus(int mode) override; 93 void SetRtxSendStatus(int mode) override;
93 int RtxSendStatus() const override; 94 int RtxSendStatus() const override;
94 95
95 void SetRtxSsrc(uint32_t ssrc) override; 96 void SetRtxSsrc(uint32_t ssrc) override;
96 97
97 void SetRtxSendPayloadType(int payload_type, 98 void SetRtxSendPayloadType(int payload_type,
98 int associated_payload_type) override; 99 int associated_payload_type) override;
99 100
101 rtc::Optional<uint32_t> FlexfecSsrc() const override;
102
100 // Sends kRtcpByeCode when going from true to false. 103 // Sends kRtcpByeCode when going from true to false.
101 int32_t SetSendingStatus(bool sending) override; 104 int32_t SetSendingStatus(bool sending) override;
102 105
103 bool Sending() const override; 106 bool Sending() const override;
104 107
105 // Drops or relays media packets. 108 // Drops or relays media packets.
106 void SetSendingMediaStatus(bool sending) override; 109 void SetSendingMediaStatus(bool sending) override;
107 110
108 bool SendingMedia() const override; 111 bool SendingMedia() const override;
109 112
(...skipping 243 matching lines...) Expand 10 before | Expand all | Expand 10 after
353 PacketLossStats receive_loss_stats_; 356 PacketLossStats receive_loss_stats_;
354 357
355 // The processed RTT from RtcpRttStats. 358 // The processed RTT from RtcpRttStats.
356 rtc::CriticalSection critical_section_rtt_; 359 rtc::CriticalSection critical_section_rtt_;
357 int64_t rtt_ms_; 360 int64_t rtt_ms_;
358 }; 361 };
359 362
360 } // namespace webrtc 363 } // namespace webrtc
361 364
362 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 365 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
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