Files
platform-external-webrtc/call/rtcp_demuxer.cc
Danil Chapovalov b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00

102 lines
3.5 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 "call/rtcp_demuxer.h"
#include "api/rtp_headers.h"
#include "call/rtcp_packet_sink_interface.h"
#include "call/rtp_rtcp_demuxer_helper.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/checks.h"
namespace webrtc {
RtcpDemuxer::RtcpDemuxer() = default;
RtcpDemuxer::~RtcpDemuxer() {
RTC_DCHECK(ssrc_sinks_.empty());
RTC_DCHECK(rsid_sinks_.empty());
RTC_DCHECK(broadcast_sinks_.empty());
}
void RtcpDemuxer::AddSink(uint32_t sender_ssrc, RtcpPacketSinkInterface* sink) {
RTC_DCHECK(sink);
RTC_DCHECK(!ContainerHasKey(broadcast_sinks_, sink));
RTC_DCHECK(!MultimapAssociationExists(ssrc_sinks_, sender_ssrc, sink));
ssrc_sinks_.emplace(sender_ssrc, sink);
}
void RtcpDemuxer::AddSink(const std::string& rsid,
RtcpPacketSinkInterface* sink) {
RTC_DCHECK(StreamId::IsLegalName(rsid));
RTC_DCHECK(sink);
RTC_DCHECK(!ContainerHasKey(broadcast_sinks_, sink));
RTC_DCHECK(!MultimapAssociationExists(rsid_sinks_, rsid, sink));
rsid_sinks_.emplace(rsid, sink);
}
void RtcpDemuxer::AddBroadcastSink(RtcpPacketSinkInterface* sink) {
RTC_DCHECK(sink);
RTC_DCHECK(!MultimapHasValue(ssrc_sinks_, sink));
RTC_DCHECK(!MultimapHasValue(rsid_sinks_, sink));
RTC_DCHECK(!ContainerHasKey(broadcast_sinks_, sink));
broadcast_sinks_.push_back(sink);
}
void RtcpDemuxer::RemoveSink(const RtcpPacketSinkInterface* sink) {
RTC_DCHECK(sink);
size_t removal_count = RemoveFromMultimapByValue(&ssrc_sinks_, sink) +
RemoveFromMultimapByValue(&rsid_sinks_, sink);
RTC_DCHECK_GT(removal_count, 0);
}
void RtcpDemuxer::RemoveBroadcastSink(const RtcpPacketSinkInterface* sink) {
RTC_DCHECK(sink);
auto it = std::find(broadcast_sinks_.begin(), broadcast_sinks_.end(), sink);
RTC_DCHECK(it != broadcast_sinks_.end());
broadcast_sinks_.erase(it);
}
void RtcpDemuxer::OnRtcpPacket(rtc::ArrayView<const uint8_t> packet) {
// Perform sender-SSRC-based demuxing for packets with a sender-SSRC.
absl::optional<uint32_t> sender_ssrc = ParseRtcpPacketSenderSsrc(packet);
if (sender_ssrc) {
auto it_range = ssrc_sinks_.equal_range(*sender_ssrc);
for (auto it = it_range.first; it != it_range.second; ++it) {
it->second->OnRtcpPacket(packet);
}
}
// All packets, even those without a sender-SSRC, are broadcast to sinks
// which listen to broadcasts.
for (RtcpPacketSinkInterface* sink : broadcast_sinks_) {
sink->OnRtcpPacket(packet);
}
}
void RtcpDemuxer::OnSsrcBoundToRsid(const std::string& rsid, uint32_t ssrc) {
// Record the new SSRC association for all of the sinks that were associated
// with the RSID.
auto it_range = rsid_sinks_.equal_range(rsid);
for (auto it = it_range.first; it != it_range.second; ++it) {
RtcpPacketSinkInterface* sink = it->second;
// Watch out for pre-existing SSRC-based associations.
if (!MultimapAssociationExists(ssrc_sinks_, ssrc, sink)) {
AddSink(ssrc, sink);
}
}
// RSIDs are uniquely associated with SSRCs; no need to keep in memory
// the RSID-to-sink association of resolved RSIDs.
rsid_sinks_.erase(it_range.first, it_range.second);
}
} // namespace webrtc