mirror of
https://github.com/mollyim/webrtc.git
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e9dad5f053
Note: google3 needs to set this clock before we can start using it. Bug: webrtc:11327 Change-Id: I0436c6633976afe208f28601fdfd50e0f6f54d6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214480 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Paul Hallak <phallak@google.com> Cr-Commit-Position: refs/heads/master@{#33653}
112 lines
3.9 KiB
C++
112 lines
3.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
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#define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
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#include <string>
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#include "api/rtp_headers.h"
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#include "api/task_queue/task_queue_base.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/ntp_time.h"
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namespace webrtc {
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class ReceiveStatisticsProvider;
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class Transport;
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// Interface to watch incoming rtcp packets by media (rtp) receiver.
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class MediaReceiverRtcpObserver {
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public:
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virtual ~MediaReceiverRtcpObserver() = default;
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// All message handlers have default empty implementation. This way users only
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// need to implement the ones they are interested in.
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virtual void OnSenderReport(uint32_t sender_ssrc,
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NtpTime ntp_time,
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uint32_t rtp_time) {}
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virtual void OnBye(uint32_t sender_ssrc) {}
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virtual void OnBitrateAllocation(uint32_t sender_ssrc,
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const VideoBitrateAllocation& allocation) {}
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};
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struct RtcpTransceiverConfig {
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RtcpTransceiverConfig();
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RtcpTransceiverConfig(const RtcpTransceiverConfig&);
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RtcpTransceiverConfig& operator=(const RtcpTransceiverConfig&);
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~RtcpTransceiverConfig();
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// Logs the error and returns false if configuration miss key objects or
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// is inconsistant. May log warnings.
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bool Validate() const;
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// Used to prepend all log messages. Can be empty.
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std::string debug_id;
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// Ssrc to use as default sender ssrc, e.g. for transport-wide feedbacks.
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uint32_t feedback_ssrc = 1;
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// Canonical End-Point Identifier of the local particiapnt.
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// Defined in rfc3550 section 6 note 2 and section 6.5.1.
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std::string cname;
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// Maximum packet size outgoing transport accepts.
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size_t max_packet_size = 1200;
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// The clock to use when querying for the NTP time. Should be set.
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Clock* clock = nullptr;
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// Transport to send rtcp packets to. Should be set.
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Transport* outgoing_transport = nullptr;
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// Queue for scheduling delayed tasks, e.g. sending periodic compound packets.
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TaskQueueBase* task_queue = nullptr;
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// Rtcp report block generator for outgoing receiver reports.
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ReceiveStatisticsProvider* receive_statistics = nullptr;
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// Callback to pass result of rtt calculation. Should outlive RtcpTransceiver.
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// Callbacks will be invoked on the task_queue.
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RtcpRttStats* rtt_observer = nullptr;
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// Configures if sending should
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// enforce compound packets: https://tools.ietf.org/html/rfc4585#section-3.1
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// or allow reduced size packets: https://tools.ietf.org/html/rfc5506
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// Receiving accepts both compound and reduced-size packets.
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RtcpMode rtcp_mode = RtcpMode::kCompound;
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//
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// Tuning parameters.
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//
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// Initial state if |outgoing_transport| ready to accept packets.
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bool initial_ready_to_send = true;
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// Delay before 1st periodic compound packet.
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int initial_report_delay_ms = 500;
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// Period between periodic compound packets.
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int report_period_ms = 1000;
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//
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// Flags for features and experiments.
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//
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bool schedule_periodic_compound_packets = true;
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// Estimate RTT as non-sender as described in
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// https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
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bool non_sender_rtt_measurement = false;
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// Allows a REMB message to be sent immediately when SetRemb is called without
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// having to wait for the next compount message to be sent.
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bool send_remb_on_change = false;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
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