rtp_demuxer.cc | Explore in Territory

/*
 *  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/rtp_demuxer.h"

#include "absl/strings/string_view.h"
#include "call/rtp_packet_sink_interface.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"

namespace webrtc {
namespace {

template <typename Container, typename Value>
size_t RemoveFromMultimapByValue(Container* multimap, const Value& value) { … }

template <typename Map, typename Value>
size_t RemoveFromMapByValue(Map* map, const Value& value) { … }

}  // namespace

RtpDemuxerCriteria::RtpDemuxerCriteria(
    absl::string_view mid,
    absl::string_view rsid /*= absl::string_view()*/)
    : … { … }

RtpDemuxerCriteria::RtpDemuxerCriteria() = default;
RtpDemuxerCriteria::~RtpDemuxerCriteria() = default;

bool RtpDemuxerCriteria::operator==(const RtpDemuxerCriteria& other) const { … }

bool RtpDemuxerCriteria::operator!=(const RtpDemuxerCriteria& other) const { … }

std::string RtpDemuxerCriteria::ToString() const { … }

// static
std::string RtpDemuxer::DescribePacket(const RtpPacketReceived& packet) { … }

RtpDemuxer::RtpDemuxer(bool use_mid /* = true*/) : … { … }

RtpDemuxer::~RtpDemuxer() { … }

bool RtpDemuxer::AddSink(const RtpDemuxerCriteria& criteria,
                         RtpPacketSinkInterface* sink) { … }

bool RtpDemuxer::CriteriaWouldConflict(
    const RtpDemuxerCriteria& criteria) const { … }

void RtpDemuxer::RefreshKnownMids() { … }

bool RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) { … }

void RtpDemuxer::AddSink(absl::string_view rsid, RtpPacketSinkInterface* sink) { … }

bool RtpDemuxer::RemoveSink(const RtpPacketSinkInterface* sink) { … }

flat_set<uint32_t> RtpDemuxer::GetSsrcsForSink(
    const RtpPacketSinkInterface* sink) const { … }

bool RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet) { … }

RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
    const RtpPacketReceived& packet) { … }

RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMid(absl::string_view mid,
                                                     uint32_t ssrc) { … }

RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByMidRsid(absl::string_view mid,
                                                         absl::string_view rsid,
                                                         uint32_t ssrc) { … }

RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByRsid(absl::string_view rsid,
                                                      uint32_t ssrc) { … }

RtpPacketSinkInterface* RtpDemuxer::ResolveSinkByPayloadType(
    uint8_t payload_type,
    uint32_t ssrc) { … }

void RtpDemuxer::AddSsrcSinkBinding(uint32_t ssrc,
                                    RtpPacketSinkInterface* sink) { … }

}  // namespace webrtc
chromium/third_party/webrtc/call/rtp_demuxer.cc
