| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_ |
| |
| #include <string> |
| |
| #include "common_types.h" // NOLINT(build/include) |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "rtc_base/task_queue.h" |
| |
| namespace webrtc { |
| class ReceiveStatisticsProvider; |
| class Transport; |
| |
| // Interface to watch incoming rtcp packets by media (rtp) receiver. |
| class MediaReceiverRtcpObserver { |
| public: |
| virtual ~MediaReceiverRtcpObserver() = default; |
| |
| // All message handlers have default empty implementation. This way user needs |
| // to implement only those she is interested in. |
| virtual void OnSenderReport(uint32_t sender_ssrc, |
| NtpTime ntp_time, |
| uint32_t rtp_time) {} |
| virtual void OnBye(uint32_t sender_ssrc) {} |
| virtual void OnBitrateAllocation(uint32_t sender_ssrc, |
| const BitrateAllocation& allocation) {} |
| }; |
| |
| struct RtcpTransceiverConfig { |
| RtcpTransceiverConfig(); |
| RtcpTransceiverConfig(const RtcpTransceiverConfig&); |
| RtcpTransceiverConfig& operator=(const RtcpTransceiverConfig&); |
| ~RtcpTransceiverConfig(); |
| |
| // Logs the error and returns false if configuration miss key objects or |
| // is inconsistant. May log warnings. |
| bool Validate() const; |
| |
| // Used to prepend all log messages. Can be empty. |
| std::string debug_id; |
| |
| // Ssrc to use as default sender ssrc, e.g. for transport-wide feedbacks. |
| uint32_t feedback_ssrc = 1; |
| |
| // Canonical End-Point Identifier of the local particiapnt. |
| // Defined in rfc3550 section 6 note 2 and section 6.5.1. |
| std::string cname; |
| |
| // Maximum packet size outgoing transport accepts. |
| size_t max_packet_size = 1200; |
| |
| // Transport to send rtcp packets to. Should be set. |
| Transport* outgoing_transport = nullptr; |
| |
| // Queue for scheduling delayed tasks, e.g. sending periodic compound packets. |
| rtc::TaskQueue* task_queue = nullptr; |
| |
| // Rtcp report block generator for outgoing receiver reports. |
| ReceiveStatisticsProvider* receive_statistics = nullptr; |
| |
| // Callback to pass result of rtt calculation. Should outlive RtcpTransceiver. |
| // Callbacks will be invoked on the task_queue. |
| RtcpRttStats* rtt_observer = nullptr; |
| |
| // Configures if sending should |
| // enforce compound packets: https://tools.ietf.org/html/rfc4585#section-3.1 |
| // or allow reduced size packets: https://tools.ietf.org/html/rfc5506 |
| // Receiving accepts both compound and reduced-size packets. |
| RtcpMode rtcp_mode = RtcpMode::kCompound; |
| // |
| // Tuning parameters. |
| // |
| // Delay before 1st periodic compound packet. |
| int initial_report_delay_ms = 500; |
| |
| // Period between periodic compound packets. |
| int report_period_ms = 1000; |
| |
| // |
| // Flags for features and experiments. |
| // |
| bool schedule_periodic_compound_packets = true; |
| // Estimate RTT as non-sender as described in |
| // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5 |
| bool non_sender_rtt_measurement = false; |
| // Copies LastSR/DelaySinceLastSR for previous report block to avoid |
| // triggering bug in older version of RtcpReceiver. |
| // TODO(bugs.webrtc.org/8805): Change to false by default then remove when |
| // all major webrtc clients updated with the fix in RtcpReceiver. |
| bool avoid_zero_last_sr_in_last_report_block = true; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_ |