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https://github.com/mollyim/webrtc.git
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3eedc90052
Move warning about missing receive_statistics to AddReceiver to avoid producing it for rtp send only endpoints. Remove warning about missing cname as unimportant. Bug: webrtc:8239 Change-Id: I8a90aa4b378284b9c68f67678b2392b9461c95b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264825 Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37093}
80 lines
2.8 KiB
C++
80 lines
2.8 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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#include "modules/rtp_rtcp/source/rtcp_transceiver_config.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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RtcpTransceiverConfig::RtcpTransceiverConfig() = default;
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RtcpTransceiverConfig::RtcpTransceiverConfig(const RtcpTransceiverConfig&) =
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default;
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RtcpTransceiverConfig& RtcpTransceiverConfig::operator=(
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const RtcpTransceiverConfig&) = default;
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RtcpTransceiverConfig::~RtcpTransceiverConfig() = default;
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bool RtcpTransceiverConfig::Validate() const {
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if (feedback_ssrc == 0) {
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RTC_LOG(LS_WARNING)
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<< debug_id
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<< "Ssrc 0 may be treated by some implementation as invalid.";
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}
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if (cname.size() > 255) {
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RTC_LOG(LS_ERROR) << debug_id << "cname can be maximum 255 characters.";
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return false;
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}
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if (max_packet_size < 100) {
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RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
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<< " is too small.";
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return false;
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}
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if (max_packet_size > IP_PACKET_SIZE) {
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RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
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<< " more than " << IP_PACKET_SIZE << " is unsupported.";
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return false;
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}
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if (clock == nullptr) {
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RTC_LOG(LS_ERROR) << debug_id << "clock must be set";
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return false;
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}
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if (!outgoing_transport) {
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RTC_LOG(LS_ERROR) << debug_id << "outgoing transport must be set";
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return false;
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}
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if (initial_report_delay < TimeDelta::Zero()) {
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RTC_LOG(LS_ERROR) << debug_id << "delay " << initial_report_delay.ms()
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<< "ms before first report shouldn't be negative.";
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return false;
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}
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if (report_period <= TimeDelta::Zero()) {
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RTC_LOG(LS_ERROR) << debug_id << "period " << report_period.ms()
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<< "ms between reports should be positive.";
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return false;
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}
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if (schedule_periodic_compound_packets && task_queue == nullptr) {
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RTC_LOG(LS_ERROR) << debug_id
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<< "missing task queue for periodic compound packets";
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return false;
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}
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if (rtcp_mode != RtcpMode::kCompound && rtcp_mode != RtcpMode::kReducedSize) {
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RTC_LOG(LS_ERROR) << debug_id << "unsupported rtcp mode";
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return false;
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}
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if (non_sender_rtt_measurement && !network_link_observer) {
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RTC_LOG(LS_WARNING) << debug_id
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<< "Enabled special feature to calculate rtt, but no "
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"rtt observer is provided.";
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}
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return true;
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}
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} // namespace webrtc
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