Add support for two-byte RTP header extensions to UsedRtpHeaderExtensionIds
Bug: webrtc:9985 Change-Id: I4c736cb920d1ac03e43c98a81218f037feebcd97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139601 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28346}
This commit is contained in:
parent
2ffcd8256f
commit
746dd0dbe6
@ -71,6 +71,7 @@ rtc_static_library("rtc_pc_base") {
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"srtp_transport.h",
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"transport_stats.cc",
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"transport_stats.h",
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"used_ids.h",
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]
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deps = [
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@ -276,6 +277,7 @@ if (rtc_include_tests) {
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"srtp_transport_unittest.cc",
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"test/rtp_transport_test_util.h",
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"test/srtp_test_util.h",
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"used_ids_unittest.cc",
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]
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include_dirs = [ "//third_party/libsrtp/srtp" ]
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@ -31,6 +31,7 @@
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#include "pc/media_protocol_names.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/srtp_filter.h"
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#include "pc/used_ids.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/helpers.h"
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#include "rtc_base/logging.h"
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@ -281,94 +282,6 @@ void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
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codecs->end());
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}
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template <typename IdStruct>
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class UsedIds {
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public:
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UsedIds(int min_allowed_id, int max_allowed_id)
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: min_allowed_id_(min_allowed_id),
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max_allowed_id_(max_allowed_id),
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next_id_(max_allowed_id) {}
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// Loops through all Id in |ids| and changes its id if it is
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// already in use by another IdStruct. Call this methods with all Id
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// in a session description to make sure no duplicate ids exists.
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// Note that typename Id must be a type of IdStruct.
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template <typename Id>
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void FindAndSetIdUsed(std::vector<Id>* ids) {
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for (const Id& id : *ids) {
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FindAndSetIdUsed(&id);
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}
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}
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// Finds and sets an unused id if the |idstruct| id is already in use.
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void FindAndSetIdUsed(IdStruct* idstruct) {
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const int original_id = idstruct->id;
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int new_id = idstruct->id;
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if (original_id > max_allowed_id_ || original_id < min_allowed_id_) {
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// If the original id is not in range - this is an id that can't be
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// dynamically changed.
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return;
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}
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if (IsIdUsed(original_id)) {
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new_id = FindUnusedId();
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RTC_LOG(LS_WARNING) << "Duplicate id found. Reassigning from "
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<< original_id << " to " << new_id;
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idstruct->id = new_id;
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}
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SetIdUsed(new_id);
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}
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private:
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// Returns the first unused id in reverse order.
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// This hopefully reduce the risk of more collisions. We want to change the
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// default ids as little as possible.
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int FindUnusedId() {
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while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) {
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--next_id_;
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}
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RTC_DCHECK(next_id_ >= min_allowed_id_);
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return next_id_;
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}
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bool IsIdUsed(int new_id) { return id_set_.find(new_id) != id_set_.end(); }
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void SetIdUsed(int new_id) { id_set_.insert(new_id); }
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const int min_allowed_id_;
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const int max_allowed_id_;
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int next_id_;
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std::set<int> id_set_;
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};
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// Helper class used for finding duplicate RTP payload types among audio, video
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// and data codecs. When bundle is used the payload types may not collide.
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class UsedPayloadTypes : public UsedIds<Codec> {
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public:
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UsedPayloadTypes()
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: UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {}
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private:
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static const int kDynamicPayloadTypeMin = 96;
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static const int kDynamicPayloadTypeMax = 127;
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};
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// Helper class used for finding duplicate RTP Header extension ids among
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// audio and video extensions. Only applies to one-byte header extensions at the
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// moment. ids > 14 will always be reported as available.
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// TODO(kron): This class needs to be refactored when we start to send two-byte
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// header extensions.
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class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {
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public:
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UsedRtpHeaderExtensionIds()
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: UsedIds<webrtc::RtpExtension>(
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webrtc::RtpExtension::kMinId,
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webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) {}
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private:
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};
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static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
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const SenderOptions& sender,
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const std::string& rtcp_cname,
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@ -1512,8 +1425,9 @@ std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
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RtpHeaderExtensions audio_rtp_extensions;
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RtpHeaderExtensions video_rtp_extensions;
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GetRtpHdrExtsToOffer(current_active_contents, &audio_rtp_extensions,
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&video_rtp_extensions);
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GetRtpHdrExtsToOffer(current_active_contents,
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session_options.offer_extmap_allow_mixed,
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&audio_rtp_extensions, &video_rtp_extensions);
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auto offer = absl::make_unique<SessionDescription>();
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@ -1958,11 +1872,19 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
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void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
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const std::vector<const ContentInfo*>& current_active_contents,
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bool extmap_allow_mixed,
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RtpHeaderExtensions* offer_audio_extensions,
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RtpHeaderExtensions* offer_video_extensions) const {
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// All header extensions allocated from the same range to avoid potential
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// issues when using BUNDLE.
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UsedRtpHeaderExtensionIds used_ids;
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// Strictly speaking the SDP attribute extmap_allow_mixed signals that the
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// receiver supports an RTP stream where one- and two-byte RTP header
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// extensions are mixed. For backwards compatibility reasons it's used in
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// WebRTC to signal that two-byte RTP header extensions are supported.
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UsedRtpHeaderExtensionIds used_ids(
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extmap_allow_mixed ? UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed
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: UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly);
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RtpHeaderExtensions all_regular_extensions;
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RtpHeaderExtensions all_encrypted_extensions;
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@ -203,6 +203,7 @@ class MediaSessionDescriptionFactory {
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RtpDataCodecs* rtp_data_codecs) const;
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void GetRtpHdrExtsToOffer(
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const std::vector<const ContentInfo*>& current_active_contents,
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bool extmap_allow_mixed,
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RtpHeaderExtensions* audio_extensions,
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RtpHeaderExtensions* video_extensions) const;
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bool AddTransportOffer(const std::string& content_name,
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168
pc/used_ids.h
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168
pc/used_ids.h
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@ -0,0 +1,168 @@
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/*
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* Copyright 2019 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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#ifndef PC_USED_IDS_H_
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#define PC_USED_IDS_H_
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#include <set>
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#include <vector>
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#include "api/rtp_parameters.h"
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#include "media/base/codec.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace cricket {
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template <typename IdStruct>
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class UsedIds {
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public:
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UsedIds(int min_allowed_id, int max_allowed_id)
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: min_allowed_id_(min_allowed_id),
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max_allowed_id_(max_allowed_id),
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next_id_(max_allowed_id) {}
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virtual ~UsedIds() {}
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// Loops through all Id in |ids| and changes its id if it is
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// already in use by another IdStruct. Call this methods with all Id
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// in a session description to make sure no duplicate ids exists.
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// Note that typename Id must be a type of IdStruct.
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template <typename Id>
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void FindAndSetIdUsed(std::vector<Id>* ids) {
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for (const Id& id : *ids) {
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FindAndSetIdUsed(&id);
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}
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}
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// Finds and sets an unused id if the |idstruct| id is already in use.
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void FindAndSetIdUsed(IdStruct* idstruct) {
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const int original_id = idstruct->id;
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int new_id = idstruct->id;
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if (original_id > max_allowed_id_ || original_id < min_allowed_id_) {
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// If the original id is not in range - this is an id that can't be
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// dynamically changed.
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return;
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}
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if (IsIdUsed(original_id)) {
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new_id = FindUnusedId();
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RTC_LOG(LS_WARNING) << "Duplicate id found. Reassigning from "
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<< original_id << " to " << new_id;
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idstruct->id = new_id;
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}
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SetIdUsed(new_id);
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}
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protected:
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bool IsIdUsed(int new_id) { return id_set_.find(new_id) != id_set_.end(); }
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const int min_allowed_id_;
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const int max_allowed_id_;
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private:
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// Returns the first unused id in reverse order.
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// This hopefully reduces the risk of more collisions. We want to change the
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// default ids as little as possible. This function is virtual and can be
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// overriden if the search for unused IDs should follow a specific pattern.
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virtual int FindUnusedId() {
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while (IsIdUsed(next_id_) && next_id_ >= min_allowed_id_) {
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--next_id_;
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}
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RTC_DCHECK(next_id_ >= min_allowed_id_);
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return next_id_;
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}
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void SetIdUsed(int new_id) {
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RTC_DCHECK(new_id >= min_allowed_id_);
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RTC_DCHECK(new_id <= max_allowed_id_);
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RTC_DCHECK(!IsIdUsed(new_id));
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id_set_.insert(new_id);
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}
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int next_id_;
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std::set<int> id_set_;
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};
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// Helper class used for finding duplicate RTP payload types among audio, video
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// and data codecs. When bundle is used the payload types may not collide.
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class UsedPayloadTypes : public UsedIds<Codec> {
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public:
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UsedPayloadTypes()
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: UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {}
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private:
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static const int kDynamicPayloadTypeMin = 96;
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static const int kDynamicPayloadTypeMax = 127;
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};
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// Helper class used for finding duplicate RTP Header extension ids among
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// audio and video extensions.
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class UsedRtpHeaderExtensionIds : public UsedIds<webrtc::RtpExtension> {
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public:
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enum class IdDomain {
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// Only allocate IDs that fit in one-byte header extensions.
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kOneByteOnly,
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// Prefer to allocate one-byte header extension IDs, but overflow to
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// two-byte if none are left.
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kTwoByteAllowed,
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};
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explicit UsedRtpHeaderExtensionIds(IdDomain id_domain)
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: UsedIds<webrtc::RtpExtension>(
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webrtc::RtpExtension::kMinId,
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id_domain == IdDomain::kTwoByteAllowed
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? webrtc::RtpExtension::kMaxId
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: webrtc::RtpExtension::kOneByteHeaderExtensionMaxId),
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id_domain_(id_domain),
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next_extension_id_(webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) {
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}
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private:
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// Returns the first unused id in reverse order from the max id of one byte
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// header extensions. This hopefully reduce the risk of more collisions. We
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// want to change the default ids as little as possible. If no unused id is
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// found and two byte header extensions are enabled (i.e.,
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// |extmap_allow_mixed_| is true), search for unused ids from 15 to 255.
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int FindUnusedId() override {
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if (next_extension_id_ <=
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webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) {
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// First search in reverse order from the max id of one byte header
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// extensions.
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while (IsIdUsed(next_extension_id_) &&
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next_extension_id_ >= min_allowed_id_) {
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--next_extension_id_;
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}
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}
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if (id_domain_ == IdDomain::kTwoByteAllowed) {
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if (next_extension_id_ < min_allowed_id_) {
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// We have searched among all one-byte IDs without finding an unused ID,
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// continue at the first two-byte ID.
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next_extension_id_ =
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webrtc::RtpExtension::kOneByteHeaderExtensionMaxId + 1;
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}
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if (next_extension_id_ >
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webrtc::RtpExtension::kOneByteHeaderExtensionMaxId) {
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while (IsIdUsed(next_extension_id_) &&
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next_extension_id_ <= max_allowed_id_) {
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++next_extension_id_;
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}
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}
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}
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RTC_DCHECK(next_extension_id_ >= min_allowed_id_);
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RTC_DCHECK(next_extension_id_ <= max_allowed_id_);
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return next_extension_id_;
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}
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const IdDomain id_domain_;
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int next_extension_id_;
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};
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} // namespace cricket
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#endif // PC_USED_IDS_H_
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176
pc/used_ids_unittest.cc
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176
pc/used_ids_unittest.cc
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@ -0,0 +1,176 @@
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/*
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* Copyright 2019 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 "pc/used_ids.h"
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#include "test/gtest.h"
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using cricket::UsedIds;
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using cricket::UsedRtpHeaderExtensionIds;
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struct Foo {
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int id;
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};
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TEST(UsedIdsTest, UniqueIdsAreUnchanged) {
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UsedIds<Foo> used_ids(1, 5);
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for (int i = 1; i <= 5; ++i) {
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Foo id = {i};
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used_ids.FindAndSetIdUsed(&id);
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EXPECT_EQ(id.id, i);
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}
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}
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TEST(UsedIdsTest, IdsOutsideRangeAreUnchanged) {
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UsedIds<Foo> used_ids(1, 5);
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Foo id_11 = {11};
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Foo id_12 = {12};
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Foo id_12_collision = {12};
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Foo id_13 = {13};
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Foo id_13_collision = {13};
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used_ids.FindAndSetIdUsed(&id_11);
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EXPECT_EQ(id_11.id, 11);
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used_ids.FindAndSetIdUsed(&id_12);
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EXPECT_EQ(id_12.id, 12);
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used_ids.FindAndSetIdUsed(&id_12_collision);
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EXPECT_EQ(id_12_collision.id, 12);
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used_ids.FindAndSetIdUsed(&id_13);
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EXPECT_EQ(id_13.id, 13);
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used_ids.FindAndSetIdUsed(&id_13_collision);
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EXPECT_EQ(id_13_collision.id, 13);
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}
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TEST(UsedIdsTest, CollisionsAreReassignedIdsInReverseOrder) {
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UsedIds<Foo> used_ids(1, 10);
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Foo id_1 = {1};
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Foo id_2 = {2};
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Foo id_2_collision = {2};
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Foo id_3 = {3};
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Foo id_3_collision = {3};
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used_ids.FindAndSetIdUsed(&id_1);
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used_ids.FindAndSetIdUsed(&id_2);
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used_ids.FindAndSetIdUsed(&id_2_collision);
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EXPECT_EQ(id_2_collision.id, 10);
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used_ids.FindAndSetIdUsed(&id_3);
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used_ids.FindAndSetIdUsed(&id_3_collision);
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EXPECT_EQ(id_3_collision.id, 9);
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}
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struct TestParams {
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UsedRtpHeaderExtensionIds::IdDomain id_domain;
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int max_id;
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};
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class UsedRtpHeaderExtensionIdsTest
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: public ::testing::TestWithParam<TestParams> {};
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constexpr TestParams kOneByteTestParams = {
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UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly, 14};
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constexpr TestParams kTwoByteTestParams = {
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UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed, 255};
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INSTANTIATE_TEST_SUITE_P(,
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UsedRtpHeaderExtensionIdsTest,
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::testing::Values(kOneByteTestParams,
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kTwoByteTestParams));
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TEST_P(UsedRtpHeaderExtensionIdsTest, UniqueIdsAreUnchanged) {
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UsedRtpHeaderExtensionIds used_ids(GetParam().id_domain);
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// Fill all IDs.
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for (int j = 1; j <= GetParam().max_id; ++j) {
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webrtc::RtpExtension extension("", j);
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used_ids.FindAndSetIdUsed(&extension);
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EXPECT_EQ(extension.id, j);
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}
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}
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TEST_P(UsedRtpHeaderExtensionIdsTest, PrioritizeReassignmentToOneByteIds) {
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UsedRtpHeaderExtensionIds used_ids(GetParam().id_domain);
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webrtc::RtpExtension id_1("", 1);
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webrtc::RtpExtension id_2("", 2);
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webrtc::RtpExtension id_2_collision("", 2);
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webrtc::RtpExtension id_3("", 3);
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webrtc::RtpExtension id_3_collision("", 3);
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// Expect that colliding IDs are reassigned to one-byte IDs.
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used_ids.FindAndSetIdUsed(&id_1);
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used_ids.FindAndSetIdUsed(&id_2);
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used_ids.FindAndSetIdUsed(&id_2_collision);
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EXPECT_EQ(id_2_collision.id, 14);
|
||||
used_ids.FindAndSetIdUsed(&id_3);
|
||||
used_ids.FindAndSetIdUsed(&id_3_collision);
|
||||
EXPECT_EQ(id_3_collision.id, 13);
|
||||
}
|
||||
|
||||
TEST_F(UsedRtpHeaderExtensionIdsTest, TwoByteIdsAllowed) {
|
||||
UsedRtpHeaderExtensionIds used_ids(
|
||||
UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed);
|
||||
|
||||
// Fill all one byte IDs.
|
||||
for (int i = 1; i < 15; ++i) {
|
||||
webrtc::RtpExtension id("", i);
|
||||
used_ids.FindAndSetIdUsed(&id);
|
||||
}
|
||||
|
||||
// Add new extensions with colliding IDs.
|
||||
webrtc::RtpExtension id1_collision("", 1);
|
||||
webrtc::RtpExtension id2_collision("", 2);
|
||||
webrtc::RtpExtension id3_collision("", 3);
|
||||
|
||||
// Expect to reassign to two-byte header extension IDs.
|
||||
used_ids.FindAndSetIdUsed(&id1_collision);
|
||||
EXPECT_EQ(id1_collision.id, 15);
|
||||
used_ids.FindAndSetIdUsed(&id2_collision);
|
||||
EXPECT_EQ(id2_collision.id, 16);
|
||||
used_ids.FindAndSetIdUsed(&id3_collision);
|
||||
EXPECT_EQ(id3_collision.id, 17);
|
||||
}
|
||||
|
||||
// Death tests.
|
||||
// Disabled on Android because death tests misbehave on Android, see
|
||||
// base/test/gtest_util.h.
|
||||
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
||||
TEST(UsedIdsDeathTest, DieWhenAllIdsAreOccupied) {
|
||||
UsedIds<Foo> used_ids(1, 5);
|
||||
for (int i = 1; i <= 5; ++i) {
|
||||
Foo id = {i};
|
||||
used_ids.FindAndSetIdUsed(&id);
|
||||
}
|
||||
Foo id_collision = {3};
|
||||
EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id_collision), "");
|
||||
}
|
||||
|
||||
using UsedRtpHeaderExtensionIdsDeathTest = UsedRtpHeaderExtensionIdsTest;
|
||||
INSTANTIATE_TEST_SUITE_P(,
|
||||
UsedRtpHeaderExtensionIdsDeathTest,
|
||||
::testing::Values(kOneByteTestParams,
|
||||
kTwoByteTestParams));
|
||||
|
||||
TEST_P(UsedRtpHeaderExtensionIdsDeathTest, DieWhenAllIdsAreOccupied) {
|
||||
UsedRtpHeaderExtensionIds used_ids(GetParam().id_domain);
|
||||
|
||||
// Fill all IDs.
|
||||
for (int j = 1; j <= GetParam().max_id; ++j) {
|
||||
webrtc::RtpExtension id("", j);
|
||||
used_ids.FindAndSetIdUsed(&id);
|
||||
}
|
||||
|
||||
webrtc::RtpExtension id1_collision("", 1);
|
||||
webrtc::RtpExtension id2_collision("", 2);
|
||||
webrtc::RtpExtension id3_collision("", GetParam().max_id);
|
||||
|
||||
EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id1_collision), "");
|
||||
EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id2_collision), "");
|
||||
EXPECT_DEATH(used_ids.FindAndSetIdUsed(&id3_collision), "");
|
||||
}
|
||||
#endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
||||
Loading…
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Reference in New Issue
Block a user