webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.cc
Danil Chapovalov f351cfffe2 Migrate RtcpTransceiver to use webrtc::TaskQueueBase instead of rtc::TaskQueue
This changes removes an extra layer of indirection
since RtcpTransceiver doesn't own TaskQueue it uses.

Bug: None
Change-Id: Ie1ef4cd8c3fb18a8e0b7ddaf0d6a319392b9e9f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126040
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30704}
2020-03-06 11:26:51 +00:00

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 == nullptr) {
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