mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
319a675318
Bug: webrtc:8239 Change-Id: Iec3d21d6297c53388bbae88611e147fe91027c83 Reviewed-on: https://webrtc-review.googlesource.com/22800 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20953}
83 lines
3.1 KiB
C++
83 lines
3.1 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#include "modules/rtp_rtcp/source/rtcp_transceiver_config.h"
|
|
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
|
|
RtcpTransceiverConfig::RtcpTransceiverConfig() = default;
|
|
RtcpTransceiverConfig::RtcpTransceiverConfig(const RtcpTransceiverConfig&) =
|
|
default;
|
|
RtcpTransceiverConfig& RtcpTransceiverConfig::operator=(
|
|
const RtcpTransceiverConfig&) = default;
|
|
RtcpTransceiverConfig::~RtcpTransceiverConfig() = default;
|
|
|
|
bool RtcpTransceiverConfig::Validate() const {
|
|
if (feedback_ssrc == 0)
|
|
RTC_LOG(LS_WARNING)
|
|
<< debug_id
|
|
<< "Ssrc 0 may be treated by some implementation as invalid.";
|
|
if (cname.empty())
|
|
RTC_LOG(LS_WARNING) << debug_id << "missing cname for ssrc "
|
|
<< feedback_ssrc;
|
|
if (cname.size() > 255) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "cname can be maximum 255 characters.";
|
|
return false;
|
|
}
|
|
if (max_packet_size < 100) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
|
|
<< " is too small.";
|
|
return false;
|
|
}
|
|
if (max_packet_size > IP_PACKET_SIZE) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "max packet size " << max_packet_size
|
|
<< " more than " << IP_PACKET_SIZE << " is unsupported.";
|
|
return false;
|
|
}
|
|
if (!outgoing_transport) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "outgoing transport must be set";
|
|
return false;
|
|
}
|
|
if (initial_report_delay_ms < 0) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "delay " << initial_report_delay_ms
|
|
<< "ms before first report shouldn't be negative.";
|
|
return false;
|
|
}
|
|
if (report_period_ms <= 0) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "period " << report_period_ms
|
|
<< "ms between reports should be positive.";
|
|
return false;
|
|
}
|
|
if (schedule_periodic_compound_packets && !task_queue) {
|
|
RTC_LOG(LS_ERROR) << debug_id
|
|
<< "missing task queue for periodic compound packets";
|
|
return false;
|
|
}
|
|
if (rtcp_mode != RtcpMode::kCompound && rtcp_mode != RtcpMode::kReducedSize) {
|
|
RTC_LOG(LS_ERROR) << debug_id << "unsupported rtcp mode";
|
|
return false;
|
|
}
|
|
if (non_sender_rtt_measurement && !rtt_observer)
|
|
RTC_LOG(LS_WARNING) << debug_id
|
|
<< "Enabled special feature to calculate rtt, but no "
|
|
"rtt observer is provided.";
|
|
// TODO(danilchap): Remove or update the warning when RtcpTransceiver supports
|
|
// send-only sessions.
|
|
if (receive_statistics == nullptr)
|
|
RTC_LOG(LS_WARNING)
|
|
<< debug_id
|
|
<< "receive statistic should be set to generate rtcp report blocks.";
|
|
return true;
|
|
}
|
|
|
|
} // namespace webrtc
|